From fea2df79f68f7810c5d76f4dccce7989e8794226 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Thu, 18 Jun 2009 16:07:18 +0000 Subject: Importing files for 1.4.26-rc3 release. git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.26-rc3@201618 f38db490-d61c-443f-a65b-d21fe96a405b --- .lastclean | 1 + .version | 1 + ChangeLog | 24806 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 24808 insertions(+) create mode 100644 .lastclean create mode 100644 .version create mode 100644 ChangeLog diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..bb95160cb --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +33 diff --git a/.version b/.version new file mode 100644 index 000000000..58f954b8b --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.4.26-rc3 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..2b2b616df --- /dev/null +++ b/ChangeLog @@ -0,0 +1,24806 @@ +2009-06-18 Leif Madsen + + * Release Asterisk 1.4.26-rc3 + +2009-06-18 15:24 +0000 [r201600] Russell Bryant + + * res/res_musiconhold.c: Fix memory corruption and leakage related + reloads of non files mode MoH classes. For Music on Hold classes + that are not files mode, meaning that we are executing an + application that will feed us audio data, we use a thread to + monitor the external application and read audio from it. This + thread also makes use of the MoH class object. In the MoH class + destructor, we used pthread_cancel() to ask the thread to exit. + Unfortunately, the code did not wait to ensure that the thread + actually went away. What needed to be done is a pthread_join() to + ensure that the thread fully cleans up before we proceed. By + adding this one line, we resolve two significant problems: 1) + Since the thread was never joined, it never fully goes away. So, + on every reload of non-files mode MoH, an unused thread was + sticking around. 2) There was a race condition here where the + application monitoring thread could still try to access the MoH + class, even though the thread executing the MoH reload has + already destroyed it. (issue #15109) Reported by: jvandal (issue + #15123) Reported by: axisinternet (issue #15195) Reported by: + amorsen (issue AST-208) + +2009-06-17 19:59 +0000 [r201450] Mark Michelson + + * main/channel.c: Change the datastore traversal in + ast_do_masquerade to use a safe list traversal. It is possible + for datastore fixup functions to remove the datastore from the + list and free it. In particular, the queue_transfer_fixup in + app_queue does this. While I don't yet know of this causing any + crashes, it certainly could. Found while discussing a separate + issue with Brian Degenhardt. + +2009-06-17 19:28 +0000 [r201423] David Vossel + + * apps/app_mixmonitor.c: StopMixMonitor race condition (not giving + up file immediately) StopMixMonitor only indicates to the + MixMonitor thread to stop writing to the file. It does not + guarantee that the recording's file handle is available to the + dialplan immediately after execution. This results in a race + condition. To resolve this, the filestream pointer is placed in a + datastore on the channel. When StopMixMonitor is called, the + datastore is retrieved from the channel and the filestream is + closed immediately before returning to the dialplan. + Documentation indicating the use of StopMixMonitor to free files + has been updated as well. (closes issue #15259) Reported by: + travisghansen Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/283/ + +2009-06-17 18:45 +0000 [r201380] David Brooks + + * channels/chan_sip.c: Checks for NULL sip_pvt pointer in + chan_sip.c->acf_channel_read() Zombie channels could be passed, + and chan_sip.c wasn't checking for it. Could crash Asterisk. Now + checking for NULL pointer. (closes issue #15330) Reported by: + okrief Tested by: dbrooks + +2009-06-17 12:03 +0000 [r200991-201261] Kevin P. Fleming + + * include/asterisk/linkedlists.h: Correct AST_LIST_APPEND_LIST + behavior when list to be appended is empty. When the list to be + appended is empty, and the list to be appended to is *not*, + AST_LIST_APPEND_LIST would actually cause the target list to + become broken, and no longer have a pointer to its last entry. + This patch fixes the problem. (reported by Stanislaw Pitucha on + the asterisk-dev mailing list) + + * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, + build_tools/cflags-devmode.xml, main/autoservice.c, main/frame.c, + apps/app_meetme.c, main/slinfactory.c, + include/asterisk/linkedlists.h, main/file.c, + include/asterisk/channel.h, include/asterisk/frame.h: Improve + support for media paths that can generate multiple frames at + once. There are various media paths in Asterisk (codec + translators and UDPTL, primarily) that can generate more than one + frame to be generated when the application calling them expects + only a single frame. This patch addresses a number of those + cases, at least the primary ones to solve the known problems. In + addition it removes the broken TRACE_FRAMES support, fixes a + number of bugs in various frame-related API functions, and cleans + up various code paths affected by these changes. + https://reviewboard.asterisk.org/r/175/ + +2009-06-16 13:25 +0000 [r200875] Eliel C. Sardanons + + * res/res_smdi.c: Show the interface name on error, if it is not + found. If the smdiport specified is not found, show the interface + name instead of '(null)'. + +2009-06-15 15:21 +0000 [r200513] Mark Michelson + + * channels/chan_sip.c: Add INFO to our allowed methods so that + endpoints know they may send it to us. AST-223 + +2009-06-12 19:06 +0000 [r200360] Mark Michelson + + * main/channel.c: Suppress a warning message and give a better + return code when generating inband ringing after a call is + answered. (closes issue #15158) Reported by: madkins Patches: + 15158.patch uploaded by mmichelson (license 60) Tested by: + madkins + +2009-06-11 22:20 +0000 [r200185] Sean Bright + + * Makefile: Backport fix for parallel build warnings from trunk + r199781. + +2009-06-11 12:12 +0000 [r200037] Leif Madsen + + * build_tools/make_version_h: Fix path for .flavor and .version. + (issue #14737) Reported by: davidw Patches: flavor.patch uploaded + by davidw (license 780) Tested by: davidw + +2009-06-10 16:08 +0000 [r199856] Sean Bright + + * include/asterisk/utils.h: __WORDSIZE is not available on all + platforms, so use sizeof(void *) instead. + +2009-06-09 Leif Madsen + + * Release Asterisk 1.4.26-rc2 + +2009-06-08 19:28 +0000 [r199626-199628] Sean Bright + + * include/asterisk/utils.h: Fix a typo in the stack size + calculation just introduced. + + * include/asterisk/utils.h: Increase the size of our thread stack + on 64 bit processors. We were setting the stack size for each + thread to 240KB regardless of architecture, which meant that in + some scenarios we actually had less available stack space on 64 + bit processors (pointers use 8 bytes instead of 4). So now we + calculate the stack size we reserve based on the platform's + __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 + bit -> 1008KB (that's right, we're ready for 128 bit processors) + Patch typed by me but written by several members of + #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes + issue #14932) Reported by: jpiszcz Patches: + 06052009_issue14932.patch uploaded by seanbright (license 71) + Tested by: seanbright + +2009-06-05 21:19 +0000 [r199297] David Vossel + + * main/pbx.c: Fixes issue with hints giving unexpected results. + Hints with two or more devices that include ONHOLD gave + unexpected results. (closes issue #15057) Reported by: + p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel + (license 671) pbx.c.1.4.patch uploaded by p (license 558) + devicestate.c.trunk.patch uploaded by p (license 671) Tested by: + p_lindheimer, dvossel Review: + https://reviewboard.asterisk.org/r/254/ + +2009-06-04 19:00 +0000 [r199138] David Vossel + + * channels/chan_iax2.c: Additional updates to AST-2009-001 + +2009-06-04 14:14 +0000 [r198957-199022] Sean Bright + + * main/asterisk.c, main/loader.c, include/asterisk.h: Safely handle + AMI connections/reload requests that occur during startup. During + asterisk startup, a lock on the list of modules is obtained by + the primary thread while each module is initialized. Issue 13778 + pointed out a problem with this approach, however. Because the + AMI is loaded before other modules, it is possible for a module + reload to be issued by a connected client (via Action: Command), + causing a deadlock. The resolution for 13778 was to move + initialization of the manager to happen after the other modules + had already been lodaded. While this fixed this particular issue, + it caused a problem for users (like FreePBX) who call AMI scripts + via an #exec in a configuration file (See issue 15189). The + solution I have come up with is to defer any reload requests that + come in until after the server is fully booted. When a call comes + in to ast_module_reload (from wherever) before we are fully + booted, the request is added to a queue of pending requests. Once + we are done booting up, we then execute these deferred requests + in turn. Note that I have tried to make this a bit more + intelligent in that it will not queue up more than 1 request for + the same module to be reloaded, and if a general reload request + comes in ('module reload') the queue is flushed and we only issue + a single deferred reload for the entire system. As for how this + will impact existing installations - Before 13778, a reload + issued before module initialization was completed would result in + a deadlock. After 13778, you simply couldn't connect to the + manager during startup (which causes problems with + #exec-that-calls-AMI configuration files). I believe this is a + good general purpose solution that won't negatively impact + existing installations. (closes issue #15189) (closes issue + #13778) Reported by: p_lindheimer Patches: + 06032009_15189_deferred_reloads.diff uploaded by seanbright + (license 71) Tested by: p_lindheimer, seanbright Review: + https://reviewboard.asterisk.org/r/272/ + + * pbx/pbx_spool.c: Fix a possible crash in pbx_spool. We were + trying to reference members of a struct that had previously been + freed. This patch makes sure that we free the struct after it has + been removed from the spooler queue. (closes issue #15072) + Reported by: garlew Patches: spool.diff uploaded by garlew + (license 376) + +2009-06-03 15:49 +0000 [r198891] David Vossel + + * main/channel.c, res/res_features.c, include/asterisk/channel.h: + Generic call forward api, ast_call_forward() The function + ast_call_forward() forwards a call to an extension specified in + an ast_channel's call_forward string. After an ast_channel is + called, if the channel's call_forward string is set this function + can be used to forward the call to a new channel and terminate + the original one. I have included this api call in both + channel.c's ast_request_and_dial() and res_feature.c's + feature_request_and_dial(). App_dial and app_queue already + contain call forward logic specific for their application and + options. (closes issue #13630) Reported by: festr Review: + https://reviewboard.asterisk.org/r/271/ + +2009-06-01 20:07 +0000 [r198665] Tilghman Lesher + + * res/res_musiconhold.c: If using the old deprecated format, a + reload would cause the class to disappear. (closes issue #14759) + Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt + uploaded by tilghman (license 14) Tested by: lmadsen + +2009-05-30 19:36 +0000 [r198370] Sean Bright + + * res/res_jabber.c: Properly terminate AMI JabberSend response + messages. The response message (either Error or Success) needs an + extra trailing \r\n after the fields to inform the client that + the message is complete. (closes issue #14876) Reported by: srt + Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright + (license 71) asterisk_14876.patch uploaded by srt (license 378) + trunk-14876-2.diff uploaded by phsultan (license 73) + +2009-05-30 03:42 +0000 [r198311] Russell Bryant + + * res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages + expired. (closes issue #14561) Reported by: cmoss28 + +2009-05-30 02:46 +0000 [r198251] Sean Bright + + * apps/app_dial.c: Treat an empty FORWARD_CONTEXT the same way we + treat a missing one. (closes issue #15056) Reported by: + p_lindheimer Patches: 05292009_bug15056.diff uploaded by + seanbright (license 71) Tested by: p_lindheimer + +2009-05-29 18:53 +0000 [r198068] Matthew Nicholson + + * main/cdr.c, main/channel.c, res/res_features.c, + include/asterisk/cdr.h: Use AST_CDR_NOANSWER instead of + AST_CDR_NULL as the default CDR disposition. This change also + involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is + used on originated channels to distinguish: them from dialed + channels. (closes issue #12946) Reported by: meral Patches: + null-cdr2.diff uploaded by mnicholson (license 96) Tested by: + mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested + by: sum + +2009-05-29 18:14 +0000 [r197998] Sean Bright + + * Makefile: Fix 'make config' target for Slackware. There was a + missing semi-colon after the echo statement in the Makefile that + was causing problems for some users. Fix suggested by reporter. + (closes issue #15225) Reported by: pdavis + +2009-05-28 23:57 +0000 [r197895] Leif Madsen + + * apps/app_mixmonitor.c: Update MixMonitor documentation. Updated + the MixMonitor documentation for the 'b' option so that it is + more obvious that you must not optimize awat the Local channel + when using this option. (issue #14829) + +2009-05-28 Leif Madsen + + * Release Asterisk 1.4.26-rc1 + +2009-05-28 15:51 +0000 [r197620] David Vossel + + * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or + not peer 'blah' is in trunk mode or not. + +2009-05-28 15:27 +0000 [r197588] Mark Michelson + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow + for media to arrive from an alternate source when responding to a + reinvite with 491. When we receive a SIP reinvite, it is possible + that we may not be able to process the reinvite immediately since + we have also sent a reinvite out ourselves. The problem is that + whoever sent us the reinvite may have also sent a reinvite out to + another party, and that reinvite may have succeeded. As a result, + even though we are not going to accept the reinvite we just + received, it is important for us to not have problems if we + suddenly start receiving RTP from a new source. The fix for this + is to grab the media source information from the SDP of the + reinvite that we receive. This information is passed to the RTP + layer so that it will know about the alternate source for media. + Review: https://reviewboard.asterisk.org/r/252 + +2009-05-28 15:21 +0000 [r197562] Eliel C. Sardanons + + * channels/chan_sip.c: Use the address we already know when + reloading a peer with nat=yes. If we already have an address for + a peer, and we are reloading the sip configuration, try to use + that address to contact the peer, instead of getting it from the + Contact. (closes issue #15194) Reported by: ibc Patches: + sip.patch uploaded by eliel (license 64) Tested by: manwe + +2009-05-28 14:49 +0000 [r197537] Mark Michelson + + * apps/app_chanspy.c, include/asterisk/audiohook.h, + main/audiohook.c: Add flags to chanspy audiohook so that audio + stays in sync. There are two flags being added to the chanspy + audiohook here. One is the pre-existing + AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that + the read and write slinfactories on the audiohook do not skew + beyond a certain tolerance. In addition, there is a new audiohook + flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, + we do not allow for a slinfactory to build up a substantial + amount of audio before flushing it. For this particular issue, + this means that the person spying on the call will hear the + conversations in real time with very little delay in the audio. + (closes issue #13745) Reported by: geoffs Patches: 13745.patch + uploaded by mmichelson (license 60) Tested by: snblitz + +2009-05-28 13:44 +0000 [r197466] Joshua Colp + + * channels/chan_sip.c: Fix a bug where the flag indicating the + presence of rport would get overwritten by the nat setting. The + presence of rport is now stored as a separate flag. Once the + dialog is setup and authenticated (or it passes through + unauthenticated) the proper nat flag is set. (closes issue + #13823) Reported by: dimas + +2009-05-27 20:12 +0000 [r197264] Sean Bright + + * Makefile: Use bash explicitly when calling + build_tools/mkpkgconfig from the Makefile. Since we use bashisms + in build_tools/mkpkgconfig, we should call on bash explicitly + when running from the Makefile, otherwise we get errors during a + 'make install.' (closes issue #15209) Reported by: seandarcy + +2009-05-27 20:07 +0000 [r197259] Olle Johansson + + * doc/asterisk-conf.txt: Typo fix + +2009-05-27 19:09 +0000 [r197194] Tilghman Lesher + + * funcs/func_cut.c: Use a different determinator on whether to + print the delimiter, since leading fields may be blank. (closes + issue #15208) Reported by: ramonpeek Patch by me, though inspired + in part by a patch from ramonpeek + +2009-05-27 16:49 +0000 [r197124] Jeff Peeler + + * main/channel.c, include/asterisk/channel.h: Fix broken attended + transfers The bridge was terminating immediately after the + attended transfer was completed. The problem was because upon + reentering ast_channel_bridge nexteventts was checked to see if + it was set and if so could possibly return AST_BRIDGE_COMPLETE. + (closes issue #15183) Reported by: andrebarbosa Tested by: + andrebarbosa, tootai, loloski + +2009-05-27 13:54 +0000 [r197024] Sean Bright + + * apps/app_queue.c: Fix handling of the 'state_interface' option of + the 'queue add member' CLI command. This change relates to + r184980, which was a backport of the state interface changes to + app_queue from trunk. trunk and all of the 1.6.x branches are not + affected. 'queue add member' allows for specifying an interface + to use for device state when adding a queue member via CLI, but + the validation code was not properly updated to reflect this + optional argument. (closes issue #15198) Reported by: loloski + Patches: 05272009_app_queue.diff uploaded by seanbright (license + 71) Tested by: loloski + +2009-05-26 18:14 +0000 [r196826] Russell Bryant + + * res/res_convert.c: Resolve a file handle leak. The frames here + should have always been freed. However, out of luck, there was + never any memory leaked. However, after file streams became + reference counted, this code would leak the file stream for the + file being read. (closes issue #15181) Reported by: jkroon + +2009-05-26 13:06 +0000 [r196657] Joshua Colp + + * contrib/scripts/safe_asterisk: Remove some bash specific stuff + from safe_asterisk. (closes issue #10812) Reported by: paravoid + Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license + 46) + +2009-05-22 13:54 +0000 [r196116] Joshua Colp + + * channels/chan_misdn.c: Fix a bug where using immediate with mISDN + caused a cause code of 16 to get sent back instead of 1 if the + 's' extension did not exist. (closes issue #12286) Reported by: + lmamane + +2009-05-21 19:04 +0000 [r195991] David Vossel + + * channels/chan_iax2.c: 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 + +2009-05-21 15:25 +0000 [r195881] Matthew Nicholson + + * main/cdr.c, res/res_features.c, include/asterisk/cdr.h: 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 + +2009-05-21 Leif Madsen + + * Release Asterisk 1.4.25 + +2009-05-13 Leif Madsen + + * Release Asterisk 1.4.25-rc1 + +2009-05-13 13:38 +0000 [r194208] Joshua Colp + + * main/rtp.c: 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 + +2009-05-13 00:52 +0000 [r194137] Tilghman Lesher + + * main/pbx.c: Fix logic for how to proceed with a single digit + extension. (closes issue #15091) Reported by: andrew Patches: + 20090512__issue15091.diff.txt uploaded by tilghman (license 14) + Tested by: andrew + +2009-05-12 22:15 +0000 [r194028] Matthew Nicholson + + * apps/app_queue.c: This change modifies app_queue to properly + generate CDR records in failure situations. This involves setting + a proper cdr disposition coresponding to the given failure + condition and ensuring the proper information is stored in the + cdr record. (closes issue #13691) Reported by: dferrer Tested by: + mnicholson (closes issue #13637) Reported by: atis Tested by: + atis + +2009-05-12 20:39 +0000 [r193955] Tilghman Lesher + + * apps/app_voicemail.c: Avoid initializing routines if the + authentication fails. Fixes a crash (RR) issue. (closes issue + #14508) Reported by: tiziano Patches: + 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license + 377) + +2009-05-12 18:18 +0000 [r193880] Mark Michelson + + * channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if + we are actually sending a SIP CANCEL. The problem was that the + hangup code was setting the invitestate too early. The result of + this was that we would always send a CANCEL request, even if it + was not an appropriate time to do so (e.g. we have not yet + received a provisional response for our INVITE). Note that this + same fix had been applied to trunk and the 1.6.X branches + starting with revision 155467. This is why you will see this + revision being blocked from those places. AST-216 + +2009-05-11 22:48 +0000 [r193755] Tilghman Lesher + + * apps/app_voicemail.c: Move 300 bytes around on the stack, to make + more room for an extension buffer. This allows more concurrent + extensions to be copied for a single voicemail, without creating + a possibility of upsetting existing users, where a dialplan could + run out of stack space where it had run fine before. + Alternatively, we could have allocated off the heap, but that is + a larger change and would have increased the chance for + instability introduced by this change. This is really solved + starting in 1.6.0.11, as the use of an ast_str buffer allows an + unlimited number of extensions (up to available memory). We + additionally create a new warning message when the buffer length + is exceeded, permitting administrators to see an issue after the + fact, whereas previously the list was silently truncated. (closes + issue #14739) Reported by: p_lindheimer Patches: + 20090417__bug14739.diff.txt uploaded by tilghman (license 14) + Tested by: p_lindheimer + +2009-05-11 19:09 +0000 [r193613] Richard Mudgett + + * channels/chan_misdn.c: Sent wrong message to clear a call we + started if the other end has not responed yet. In the state + MISDN_CALLING (i.e. SETUP was sent but no answer has arrived + yet), it is not allowed to clear the call with RELEASE_COMPLETE. + It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only + allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, + 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. + JIRA ABE-1862 + +2009-05-11 17:35 +0000 [r193544] Leif Madsen + + * funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI + documentation. (issue #15073) Reported by: pkempgen Patches: + 20090511__issue15073.diff.txt uploaded by tilghman (license 14) + +2009-05-08 21:01 +0000 [r193391] Matthew Nicholson + + * main/channel.c: Set the proper disposition on originated calls. + (closes issue #14167) Reported by: jpt Patches: + call-file-missing-cdr2.diff uploaded by mnicholson (license 96) + Tested by: dlotina, rmartinez, mnicholson + +2009-05-08 14:51 +0000 [r193262] David Vossel + + * channels/misdn_config.c: "misdn show config" segfaults asterisk, + if no MSN lists (closes issue #14976) Reported by: alecdavis + Patches: misdn_config.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis, FabienToune + +2009-05-08 14:03 +0000 [r193193] Kevin P. Fleming + + * configs/logger.conf.sample, main/logger.c: Make absolute paths + for logger channels work properly (Note: This is not a new + feature, it was previously undocumented and broken.) The Asterisk + logger has a feature to support absolute pathnames for logger + channels, but the code implementing the feature was broken. This + has been fixed, and the absolute path feature is now documented + in the sample logger.conf. + +2009-05-07 23:41 +0000 [r193119] Tilghman Lesher + + * main/pbx.c: Fix Background within a Macro for FreePBX. If the + single digit DTMF is an extension in the specified context, then + go there and signal no DTMF. Otherwise, we should exit with that + DTMF. If we're in Macro, we'll exit and seek that DTMF as the + beginning of an extension in the Macro's calling context. If + we're not in Macro, then we'll simply seek that extension in the + calling context. Previously, someone complained about the + behavior as it related to the interior of a Gosub routine, and + the fix (#14011) inadvertently broke FreePBX (#14940). This + change should fix both of these situations, but with the possible + incompatibility that if a single digit extension does not exist + (but a longer extension COULD have matched), it would have + previously gone immediately to the "i" extension, but will now + need to wait for a timeout. (closes issue #14940) Reported by: + p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by + tilghman (license 14) Tested by: p_lindheimer + +2009-05-07 22:17 +0000 [r193050] Richard Mudgett + + * channels/chan_misdn.c: Give a more helpful message when an + incoming call's dialed extension does not match. Added the dialed + extension and context to the chan_misdn messages warning that the + dialed number cannot be matched in the dialplan. + +2009-05-07 16:29 +0000 [r192932] Tilghman Lesher + + * channels/chan_sip.c: Eliminate repetition of fullcontact during + reconstruction. If the fullcontact field appears in both the + sippeers and the sipregs table, then during reconstruction of the + field, it will otherwise be doubled. (closes issue #14754) + Reported by: Alexei Gradinari Patches: + 20090506__bug14754.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen + +2009-05-06 22:15 +0000 [r192858] Jeff Peeler + + * res/res_features.c: Make ParkedCall application stop execution of + the dialplan after hang up Just changed park_exec to always + return non-zero. I really wasn't entirely sure at first if this + was a bug. Decided it was since it would be surprising when not + using ParkedCall in the dialplan to hang up and have dialplan + execution continue. (closes issue #14555) Reported by: + francesco_r + +2009-05-06 13:30 +0000 [r192633] Joshua Colp + + * channels/chan_sip.c: Update some old logic to stop both begin and + end DTMF frames from reaching the core if rfc2833 is not enabled. + (closes issue #15036) Reported by: dimas Patches: v1-15036.patch + uploaded by dimas (license 88) + +2009-05-05 19:56 +0000 [r192524] Sean Bright + + * static-http/astman.js: Fix Javascript error when using astman.js + in Internet Explorer. Internet Explorer (tested with 7.0) does + not like trailing commas on constructs like object initializers, + so get rid of them to avoid some errors. (closes issue #15026) + Reported by: rajnishgiri Patches: bug15026.patch uploaded by + seanbright (license 71) Tested by: seanbright + +2009-05-05 18:22 +0000 [r192429-192454] Joshua Colp + + * res/res_features.c: Fix an incorrect assumption that certain + values on the channel will always exist when they may not. The + CDR code involved with bridges wrongly assumed that the currently + executing application and data values will always exist. It is + possible for this to be false when call forwarding is involved. + (closes issue #14984) Reported by: gincantalupo + + * apps/app_followme.c: Fix a bug where the followme application + would continue trying numbers after the caller hung up. (closes + issue #13624) Reported by: sgenyuk + +2009-05-04 22:37 +0000 [r192213] David Vossel + + * channels/chan_iax2.c: global mohinterpret setting is ignored + mohinterpret and mohsuggest global variables were not copied over + during build_users and build_peers. (closes issue #14728) + Reported by: dimas Patches: v1-14728.patch uploaded by dimas + (license 88) Tested by: dimas, dvossel + +2009-05-02 18:48 +0000 [r191628-191778] Mark Michelson + + * apps/app_voicemail.c: Fix a bug which resulted from the Hebrew + voicemail commit. This fixes a case where a certain message could + get played twice. (closes issue #13155) Reported by: + greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded + by greenfieldtech (license 369) Tested by: greenfieldtech + + * apps/app_chanspy.c: Kevin has informed me that thi sort of thing + is not necessary. + + * apps/app_chanspy.c: Move static buffers to outside for loops in + app_chanspy. Similar to seanbright's commit 191422, this moves + some static buffers to be defined outside of for loops since it + is undefined if memory will be re-used or if the stack will grow + with each iteration of the loop. + +2009-05-01 20:00 +0000 [r191559] Tilghman Lesher + + * channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or + 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: + causepatch uploaded by BigJimmy (license 371) + +2009-05-01 17:40 +0000 [r191488] Jeff Peeler + + * main/channel.c: Fix DTMF not being sent to other side after a + partial feature match This fixes a regression from commit 176701. + The issue was that ast_generic_bridge never exited after the + feature digit timeout had elapsed, which prevented the queued + DTMF from being sent to the other side. This issue was reported + to me directly. + +2009-05-01 15:42 +0000 [r191422] Sean Bright + + * apps/app_queue.c: Move the defintion of the a couple arrays out + of loops. According to Kevin, it is unspecified as to whether a + variable defined inside a block is allocated once by the compiler + or for each pass through the block (loops being the only + interesting case), so just define these before we get into our + loop to be sure. + +2009-04-29 23:10 +0000 [r191220] Tilghman Lesher + + * channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to + compile with FDLEAK checking enabled. + +2009-04-29 18:07 +0000 [r191096] David Brooks + + * pbx/pbx_config.c: Patch to fix tab-completion crash on "remove + extension" This patch simply removes some old code back before + Asterisk used editline. This fixes the crash that occurred when + tab-completing "remove extension". (closes issue #14689) Reported + by: isaacgal + +2009-04-29 15:23 +0000 [r191041] Sean Bright + + * apps/app_queue.c: Fix a crash in app_queue with very long member + lists. A user reported via #asterisk that with very long lists of + members, a crash occurs in ast_strdupa, so just use a single + buffer and ast_copy_string instead of stack allocating copys of + each interface name. + +2009-04-27 19:29 +0000 [r190721] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent + line endings' when autoconf 2.63 is used Attempt to make + configure script regeneration 'safe' using autoconf 2.63, which + embeds a bare CR into the script, thus making Subversion complain + about inconsistent line endings This commit changes the MIME type + of the configure script to be 'binary' thus making Subversion no + longer inspect line endings, and as a bonus 'svn diff' will no + longer try to generate diff output for it, which is not generally + useful anyway. + +2009-04-27 19:03 +0000 [r190661-190662] Russell Bryant + + * res/res_smdi.c: Fix a typo from 190661. + + * res/res_smdi.c: Resolve a crash in res_smdi when used with + chan_dahdi. When chan_dahdi goes to get an SMDI message, it + provides no search criteria. It just grabs the next message that + arrives. This code was written with the SMDI dialplan functions + in mind, since that is now the preferred method of using SMDI. + However, this broke support of it being used from chan_dahdi. + (closes AST-212) + +2009-04-23 21:07 +0000 [r190356] Russell Bryant + + * channels/chan_sip.c: Remove a bogus ast_channel_unlock(). + +2009-04-23 19:13 +0000 [r190286] Joshua Colp + + * channels/chan_local.c: Fix a bug in chan_local glare hangup + detection. If both sides of a Local channel were hung up at + around the same time it was possible for one thread to destroy + the local private structure and have the other thread immediately + try to remove the already freed structure from the local channel + list. + +2009-04-23 10:07 +0000 [r190187] Olle Johansson + + * include/asterisk/lock.h: unistd.h is required for usleep() on + Darwin. It will not hurt to include it always on other platforms + either. + +2009-04-22 21:35 +0000 [r190092] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Detect availability of + pthread_rwlock_timedwrlock() before using it. (closes issue + #14930) Reported by: tilghman Patches: + 20090420__bug14930.diff.txt uploaded by tilghman (license 14) + Tested by: mvanbaak, tilghman + +2009-04-22 19:20 +0000 [r189991] Jeff Peeler + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/chan_h323.h: Make chan_h323 respect packetization + settings Previously, packetization settings were ignored and now + they are not. A new config option 'autoframing' has been added to + mirror the way chan_sip handles it. Turning on the autoframing + option (available both as a global option or per peer) overrides + the local settings with the remote packetization settings. + Testing was performed with varying packetization levels with the + following codecs: ulaw, alaw, gsm, and g729. (closes issue + #12415) Reported by: pj Patches: + 2009012200_h323packetization.diff.txt uploaded by mvanbaak + (license 7), modified by me + +2009-04-22 14:29 +0000 [r189849] Michiel van Baak + + * contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove + \r from downloaded file On some systems, sed does not recognize + \r in the pattern the way it was used here. Use tr instead + because this works the same across systems. (closes issue #14936) + Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded + by mvanbaak (license 7) Tested by: leobrown, mvanbaak + +2009-04-21 15:52 +0000 [r189601-189664] Doug Bailey + + * utils/muted.c: Remove daemon call on systems that do not support + forking. + + * main/config.c, configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, configure.ac: Add check in configure + script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This + allows config.c to compile when linked against uclibc that does + not support these parameters + +2009-04-20 22:02 +0000 [r189537] Tilghman Lesher + + * funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for + func_odbc/ARRAY() for problems that occur with certain special + characters. In certain cases, due to the way Set() works in 1.4, + values may not get set properly. This is a workaround for 1.4 + only that corrects for these issues, without making func_odbc + more difficult to use properly. (closes issue #14614) Reported + by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by + tilghman (license 14) + double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff + uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman + +2009-04-20 21:10 +0000 [r189463-189465] Terry Wilson + + * apps/app_dial.c: Update CDR appropriately when + AST_CAUSE_NO_ANSWER is set + + * apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL + +2009-04-20 20:58 +0000 [r189462] Sean Bright + + * pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints + in AEL. AEL was not handling the case of a device hint containing + an @ symbol, which caused parking hints (e.g. + hint(park:exten@context)) to error out the parser. This patch + makes AEL treat the @ the same way it treats colon and ampersand + now, meaning the characters are included in verbatim. (closes + issue #14941) Reported by: bpgoldsb Patches: bug14941.patch + uploaded by seanbright (license 71) Tested by: bpgoldsb + +2009-04-20 19:10 +0000 [r189391] Doug Bailey + + * main/manager.c, main/db1-ast/recno/rec_open.c, + channels/chan_iax2.c: Clean up problem with manager + implementation of mmap where it was not testing against + MAP_FAILED response. Got rid of shadowed variable used in + processign the mmap results. Change test of mmap results to + compare against MAP_FAILED + +2009-04-20 14:04 +0000 [r189277] Mark Michelson + + * main/channel.c: Move the check for chan->fdno == -1 to after the + zombie/hangup check. Many users were finding that their hung up + channels were staying up and causing 100% CPU usage. (issue + #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch + uploaded by mmichelson (license 60) Tested by: falves11, bamby + +2009-04-18 01:27 +0000 [r189203] David Vossel + + * channels/chan_agent.c: Fixed autologoff in agents.conf not + working when agent logs in via AgentLogin app An agent logs in by + calling an extension that calls the AgentLogin app. In + agents.conf ackcall=always is set, so when they get a call they + have the choice to either acknowledge it or ignore it. + autologoff=10 is set as well, so if the agent ignores the call + over 10sec one may assume that the agent should be logged out + (and in this case hungup on as well), but this was not happening. + (closes issue #14091) Reported by: evandro Patches: + autologoff.diff uploaded by dvossel (license 671) Review: + http://reviewboard.digium.com/r/225/ + +2009-04-17 21:27 +0000 [r189134] Richard Mudgett + + * channels/misdn/isdn_lib.c: Modifed/added some debug messages. + JIRA ABE-1835 + +2009-04-17 15:43 +0000 [r189009] Matthew Nicholson + + * main/pbx.c: Make Busy() application set the CDR disposition to + BUSY. (closes issue #14306) Reported by: cristiandimache + +2009-04-17 14:41 +0000 [r188937-188946] Joshua Colp + + * channels/chan_sip.c: Fix a bug where a value used to create the + channel name was bogus. This commit fixes the scenario where an + incoming call is authenticated using a peer entry. Previously the + channel name was created using either the username setting from + the sip.conf entry or the IP address that the call came from. Now + the channel name will be created using the peer name itself. This + commit will not change the way the channel name is generated for + users or friends. (closes issue #14256) Reported by: Nick_Lewis + Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) + Tested by: Nick_Lewis, file + + * channels/chan_dahdi.c: Fix a situation where the DAHDI channel + private structure lock was not unlocked when it should have been. + (issue AST-210) + +2009-04-16 21:41 +0000 [r188835] Tilghman Lesher + + * channels/chan_sip.c: Only update realtime, if global option + rtupdate != false (closes issue #14885) Reported by: deepesh + Patches: 20090413__bug14885.diff.txt uploaded by tilghman + (license 14) Tested by: deepesh + +2009-04-16 21:37 +0000 [r188833] Richard Mudgett + + * channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is + enabled. Leave jitter setting alone. JIRA ABE-1835 + +2009-04-16 21:02 +0000 [r188773] Tilghman Lesher + + * apps/app_voicemail.c: Umask should not be exported into global + namespace. (closes issue #14912) Reported by: jcapp + +2009-04-15 22:08 +0000 [r188646] David Vossel + + * channels/chan_dahdi.c: National prefix inserted even when caller + ID not available When the caller ID is restricted, the expected + behavior is for the caller id to be blank. In chan_dahdi, the + national prefix is placed onto the callers number even if its + restricted (empty) causing the caller id to be the national + prefix rather than blank. (closes issue #13207) Reported by: + shawkris Patches: national_prefix.diff uploaded by dvossel + (license 671) Review: http://reviewboard.digium.com/r/220/ + +2009-04-15 20:04 +0000 [r188582] Mark Michelson + + * main/file.c: Update ast_readvideo_callback to match + ast_readaudio_callback. This fixes potential refcount errors that + may occur on ast_filestreams. AST-208 + +2009-04-14 15:02 +0000 [r188287] David Vossel + + * main/audiohook.c: audio_audiohook_write_list() does not correctly + update sample size after ast_translate. + audio_audiohook_write_list() does not take into account that the + sample size may change after translation depending on if the + original frame is is 8khz or 16khz. While no 16kz codecs are + supported in 1.4 at the moment, this will save headaches in the + future if they ever are. the sample size is now updated after + translating to reflect this possibility. Thanks to jcolp and + mmichelson for helping me work this out. (issue AST-197) + +2009-04-13 23:04 +0000 [r188149] Tilghman Lesher + + * res/res_odbc.c: If fileconfig limit exceeds our maximum, then set + the limit to the maximum. (Closes issue #14888) Reported by: + falves11 + +2009-04-10 22:16 +0000 [r187962] Jeff Peeler + + * channels/Makefile: Fix module embedding for chan_h323. Include + libchanh323.a in the modules.link file so that all the symbols + can be resolved at link time. (closes issue #11966) Reported by: + dome + +2009-04-10 19:26 +0000 [r187865] Russell Bryant + + * channels/chan_dahdi.c: Support "signaling" in addition to + "signalling". The sample configuration file has references to + both spellings. + +2009-04-10 17:28 +0000 [r187763] Tilghman Lesher + + * contrib/scripts/realtime_pgsql.sql, + contrib/scripts/sip-friends.sql: Add lastms column to the + contributed table designs + +2009-04-09 18:51 +0000 [r187484] Mark Michelson + + * channels/chan_sip.c: Handle a SIP race condition (reinvite before + an ACK) properly. RFC 5047 explains the proper course of action + to take if a reINVITE is received before the ACK from a previous + invite transaction. What we are to do is to treat the reINVITE as + if it were both an ACK and a reINVITE and process it normally. + Later, when we receive the ACK we had been expecting, we will + ignore it since its CSeq is less than the current iseqno of the + sip_pvt representing this dialog. (closes issue #13849) Reported + by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson + (license 60) Tested by: mmichelson, klaus3000 + +2009-04-09 18:39 +0000 [r187209-187482] Tilghman Lesher + + * include/asterisk/lock.h: Oops, typo + + * main/manager.c, include/asterisk/lock.h: Race condition between + ast_cli_command() and 'module unload' could cause a deadlock. Add + lock timeouts to avoid this potential deadlock. (closes issue + #14705) Reported by: jamessan Patches: + 20090320__bug14705.diff.txt uploaded by tilghman (license 14) + Tested by: jamessan + + * channels/chan_sip.c, apps/app_sendtext.c: Permit zero-length text + messages in SIP. (Related to an issue posted to the -users list, + subject "AEL2, BASE64_DECODE and hexadecimal") + + * main/astfd.c (added): Oops, missed this file in the last commit. + + * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, + utils/Makefile, include/asterisk.h, main/Makefile, main/file.c: + Add debugging mode for diagnosing file descriptor leaks. (Related + to issue #14625) + + * main/manager.c: Backport resolution for file descriptor leak in + 1.6.0 to 1.4. This fixes short reads in http manager sessions, + such as those done by the ast-gui branch. (Fixes AST-198) + +2009-04-08 19:16 +0000 [r186832-187135] Mark Michelson + + * apps/app_dial.c: Fix a crash due to too few arguments to + RetryDial. (closes issue #14852) Reported by: junky Patches: + retry_fix.diff uploaded by junky (license 177) + + * res/res_musiconhold.c: Fix a small logical error when loading moh + classes. We were unconditionally incrementing the number of + mohclasses registered. However, we should actually only increment + if the call to moh_register was successful. While this probably + has never caused problems, I noticed it and decided to fix it + anyway. + + * main/channel.c: Make a couple of changes with regards to a new + message printed in ast_read(). "ast_read() called with no + recorded file descriptor" is a new message added after a bug was + discovered. Unfortunately, it seems there are a bunch of places + that potentially make such calls to ast_read() and trigger this + error message to be displayed. This commit does two things to + help to make this message appear less. First, the message has + been downgraded to a debug level message if dev mode is not + enabled. The message means a lot more to developers than it does + to end users, and so developers should take an effort to be sure + to call ast_read only when a channel is ready to be read from. + However, since this doesn't actually cause an error in operation + and is not something a user can easily fix, we should not spam + their console with these messages. Second, the message has been + moved to after the check for any pending masquerades. ast_read() + being called with no recorded file descriptor should not + interfere with a masquerade taking place. This could be seen as a + simple way of resolving issue #14723. However, I still want to + try to clear out the existing ways of triggering this message, + since I feel that would be a better resolution for the issue. + + * formats/format_wav.c, formats/format_wav_gsm.c: Fix a few typos + of the word "frequency." (closes issue #14842) Reported by: + jvandal Patches: frequency-typo.diff uploaded by jvandal (license + 413) + + * main/channel.c: Set the AST_FEATURE_WARNING_ACTIVE flag when a + p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, + warning sounds will not be properly played to either party of the + bridge. (closes issue #14845) Reported by: adomjan + +2009-04-07 22:16 +0000 [r186775] Tilghman Lesher + + * apps/app_macro.c: Fix Macro documentation to match current (and + intended) behavior. (See -dev mailing list) + +2009-04-07 20:43 +0000 [r186719] Mark Michelson + + * main/manager.c: Ensure that \r\n is printed after the ActionID in + an OriginateResponse. (closes issue #14847) Reported by: kobaz + +2009-04-06 13:54 +0000 [r186565] Mark Michelson + + * apps/app_voicemail.c: Revert commit 186445 because it causes the + build to fail when IMAP_STORAGE is used. + +2009-04-03 20:19 +0000 [r186458] Kevin P. Fleming + + * channels/chan_dahdi.c: Fix a bug where DAHDI/Zaptel channels + would not properly switch formats when requested Don't offer + AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could + provide a slight performance benefit, the translation core in + Asterisk has some flaws when a channel driver offers multiple raw + formats. this fix is much simpler than fixing the translation + core to solve that issue (although that will be done later). + +2009-04-03 19:56 +0000 [r186415-186445] Tilghman Lesher + + * apps/app_voicemail.c: Found a conflict in the last commit, due to + multiple targets + + * apps/app_voicemail.c, configs/voicemail.conf.sample: Distinguish + in a sent email between simple sends and forwards. (closes issue + #11678) Reported by: jamessan Patches: + 20090330__bug11678.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman, lmadsen + +2009-04-03 15:48 +0000 [r186320] Joshua Colp + + * include/asterisk/crypto.h: Fix a problem with the crypto variable + definitions not actually being defined properly. (closes issue + #14804) Reported by: jvandal + +2009-04-03 01:57 +0000 [r186229] Russell Bryant + + * cdr/cdr_radius.c: Fix a memory leak in cdr_radius. I came across + this while doing some testing of my ast_channel_ao2 branch. After + running a test overnight that generated over 5 million calls, + Asterisk had taken up about 1 GB of my system memory. So, I + re-ran the test with MALLOC_DEBUG turned on. However, it showed + no leaks in Asterisk during the test, even though Asterisk was + still consuming it somehow. Instead, I turned to valgrind, which + when run with --leak-check=full, told me exactly where the leak + came from, which was from allocations inside the radiusclient-ng + library. This explains why MALLOC_DEBUG did not report it. After + a bit of analysis, I found that we were leaking a little bit of + memory every time a CDR record was passed to cdr_radius. I don't + actually have a radius server set up to receive CDR records. + However, I always have my development systems compile and install + all modules. In addition to making sure there are not build + errors across modules, always loading modules helps find bugs + like this, too, so it is strongly recommend for all developers. + +2009-04-02 21:55 +0000 [r186174] Mark Michelson + + * configs/features.conf.sample: Fix instructions in one-step + parking comment to make more sense. Changed a capital K to a + lowercase k. + +2009-04-02 17:21 +0000 [r186081] Kevin P. Fleming + + * channels/chan_dahdi.c: ensure that the buffer passed to + DAHDI_SET_BUFINFO is fully initialized + +2009-04-02 17:09 +0000 [r186057-186059] Tilghman Lesher + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 186056 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) + | 2 lines Fix for AST-2009-003 ........ + + * channels/chan_sip.c: Avoid multiple warning messages in SIP, due + to this column not existing + +2009-04-02 13:43 +0000 [r185952] Kevin P. Fleming + + * channels/chan_dahdi.c: the DAHDI_GETCONF, DAHDI_SETCONF and + DAHDI_GET_PARAMS ioctls were recently corrected to show that they + do, in fact, read data from userspace as part of their work. due + to this fix, valgrind now reports a number of cases where + chan_dahdi passed an uninitialized (or partially) buffer to these + ioctls, which could lead to unexpected behavior. this patch + corrects chan_dahdi to ensure that buffers passed to these ioctls + are always fully initialized. + +2009-04-01 19:02 +0000 [r185845] David Vossel + + * channels/chan_sip.c: Fixes issue with dropped calles due to + re-Invite glare and re-Invites never executing after a 491 + Acknowledgement for 491 responses were never being processed + because it didn't match our pending invite's seqno. Since the ACK + was never processed, the 491 frame would continue to be + retransmitted until eventually the call was dropped due to max + retries. Now during a pending invite, if we receive another + invite, we send an 491 and hold on to that glare invite's seqno + in the "glareinvite" variable for that sip_pvt struct. When ACK's + are received, we first check to see if it is in response to our + pending invite, if not we check to see if it is in response to a + glare invite. In this case, it is in response to the glare invite + and must be dealt with or the call is dropped. I've changed the + wait time for resending the re-Invite after receving a 491 + response to comply with RFC 3261. Before this patch the scheduled + re-Invite would only change a flag indicating that the re-Invite + should be sent out, now it actually sends it out as well. (closes + issue #12013) Reported by: alx Review: + http://reviewboard.digium.com/r/213/ + +2009-04-01 13:47 +0000 [r185771] Russell Bryant + + * main/channel.c: Fix a case where DTMF could bypass audiohooks. + This change fixes a situation where an audiohook that wants DTMF + would not actually get it. This is in the code path where we end + DTMF digit length emulation while handling a NULL frame. + +2009-03-31 22:00 +0000 [r185468-185599] Mark Michelson + + * apps/app_queue.c: Fix crash that would occur if an empty member + was specified in queues.conf. (closes issue #14796) Reported by: + pida + + * channels/chan_sip.c: Use AST_SCHED_DEL_SPINLOCK instead of + manually using the logic. + + * apps/app_voicemail.c: Fix Russian voicemail intro to say the word + "messages" properly. (closes issue #14736) Reported by: chappell + Patches: voicemail_no_messages.diff uploaded by chappell (license + 8) + +2009-03-31 16:37 +0000 [r185362] David Brooks + + * channels/chan_gtalk.c: Fix incorrect parsing in chan_gtalk when + xmpp contains extra whitespaces To drill into the xmpp to find + the capabilities between channels, chan_gtalk calls iks_child() + and iks_next(). iks_child() and iks_next() are functions in the + iksemel xml parsing library that traverse xml nodes. The bug here + is that both iks_child() and iks_next() will return the next + iks_struct node *regardless* of type. chan_gtalk expects the next + node to be of type IKS_TAG, which in most cases, it is, but in + this case (a call being made from the Empathy IM client), there + exists iks_struct nodes which are not IKS_TAG data (they are + extraneous whitespaces), and chan_gtalk doesn't handle that case, + so capabilities don't match, and a call cannot be made. + iks_first_tag() and iks_next_tag(), on the other hand, will not + return the very next iks_struct, but will check to see if the + next iks_struct is of type IKS_TAG. If it isn't, it will be + skipped, and the next struct of type IKS_TAG it finds will be + returned. This assures that chan_gtalk will find the iks_struct + it is looking for. This fix simply changes all calls to + iks_child() and iks_next() to become calls to iks_first_tag() and + iks_next_tag(), which resolves the capability matching. The + following is a payload listing from Empathy, which, due to the + extraneous whitespace, will not be parsed correctly by iksemel: + + + + +2009-03-31 15:34 +0000 [r185298] Mark Michelson + + * apps/app_queue.c: Fix some state_interface stuff that was in + trunk but not in the backport to 1.4. Issue #14359 was fixed + between the time that I posted the review of the backport of the + state interface change for 1.4. This merges the changes from that + issue back into 1.4. (closes issue #14359) Reported by: + francesco_r + +2009-03-31 14:06 +0000 [r185196] Joshua Colp + + * main/audiohook.c: Fix crash when moving audiohooks between + channels. Handle the scenario where we are called to move + audiohooks between channels and the source channel does not + actually have any on it. (closes issue #14734) Reported by: + corruptor + +2009-03-30 20:40 +0000 [r185120-185121] Richard Mudgett + + * channels/misdn_config.c, configs/misdn.conf.sample: Update the + channel allocation method documentation. + + * channels/misdn/isdn_lib.c: Make chan_misdn BRI TE side normally + defer channel selection to the NT side. Channel allocation + collisions are not handled by chan_misdn very well. This patch + simply avoids the problem for BRI only. For PRI, allocation + collisions are still possible but less likely since there are + simply more channels available and each end could use a different + allocation strategy. misdn.conf options available: + te_choose_channel - Use to force the TE side to allocate + channels. method - Specify the channel allocation strategy. + (closes issue #13488) Reported by: Christian_Pinedo Patches: + isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, + festr + +2009-03-30 16:17 +0000 [r184980-185031] Mark Michelson + + * apps/app_queue.c: 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/ + + * configs/queues.conf.sample, apps/app_queue.c: 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 + +2009-03-30 14:35 +0000 [r184947] Joshua Colp + + * channels/chan_sip.c: 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/ + +2009-03-29 05:51 +0000 [r184842] Russell Bryant + + * apps/app_followme.c: Ensure targs variable is fully initialized. + (closes issue #14758) Reported by: tim_ringenbach + +2009-03-27 13:06 +0000 [r184565] Joshua Colp + + * channels/chan_sip.c: 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 + +2009-03-26 22:17 +0000 [r184447] Kevin P. Fleming + + * sounds/Makefile: use new, improved 8kHz prompts + +2009-03-26 21:07 +0000 [r184388] David Vossel + + * apps/app_test.c: 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 + +2009-03-25 14:12 +0000 [r184188] Eliel C. Sardanons + + * main/asterisk.c: 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 + +2009-03-24 22:34 +0000 [r184078] Mark Michelson + + * apps/app_senddtmf.c: 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. + +2009-03-24 15:25 +0000 [r183913] Tilghman Lesher + + * configs/voicemail.conf.sample: Additionally note that the + operator option needs an 'o' extension. (Related to issue #14731) + +2009-03-23 17:59 +0000 [r183700] Mark Michelson + + * res/res_monitor.c: 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. + +2009-03-20 16:53 +0000 [r183559] Russell Bryant + + * channels/chan_iax2.c: Fix a crash in IAX2 registration handling + found during load testing with dvossel. + +2009-03-19 23:37 +0000 [r183481] Terry Wilson + + * apps/app_dial.c: Add missing datastore inherit (exists in all + other branches) + +2009-03-19 19:40 +0000 [r183386] David Vossel + + * include/asterisk/features.h, apps/app_dial.c, res/res_features.c: + 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 + +2009-03-19 19:21 +0000 [r183319-183342] Tilghman Lesher + + * channels/chan_dahdi.c: Reordering, to change prior to unlocking + + * channels/chan_dahdi.c: 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 + +2009-03-19 18:28 +0000 [r183291] Jason Parker + + * main/asterisk.exports: Export some more required symbols. + +2009-03-19 17:52 +0000 [r183145-183241] Russell Bryant + + * main/loader.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: Remove the use of RTLD_NOLOAD, as it is not + behaving like expected. + + * main/asterisk.exports: Allow the AES API to work. + + * main/asterisk.exports: Add missing semicolon in exports script. + +2009-03-19 16:15 +0000 [r183126] David Vossel + + * include/asterisk/features.h, apps/app_dial.c, res/res_features.c, + res/res_features.exports: 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/ + +2009-03-19 16:13 +0000 [r183123] Russell Bryant + + * main/asterisk.exports: Allow the CallerID API to work again. + +2009-03-19 16:04 +0000 [r183115] Mark Michelson + + * channels/chan_sip.c: 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 + +2009-03-18 20:02 +0000 [r182963-182965] Jeff Peeler + + * configure, autoconf/ast_check_openh323.m4: fix typo which broke + configure + + * channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx, + configure, autoconf/ast_check_openh323.m4, + channels/h323/compat_h323.h, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: 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) + +2009-03-18 11:31 +0000 [r182882] Kevin P. Fleming + + * include/asterisk/callerid.h, channels/chan_dahdi.c, + main/callerid.c: fix another symbol namespace issue (reported by + Andrew on asterisk-dev) + +2009-03-18 02:09 +0000 [r182810] Russell Bryant + + * main/poll.c, main/io.c, main/channel.c, main/manager.c, + channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, + include/asterisk/poll-compat.h, channels/chan_alsa.c, + main/asterisk.c, apps/app_nbscat.c, main/Makefile, + include/asterisk/autoconfig.h.in, configure.ac, main/utils.c, + include/asterisk/io.h, include/asterisk/channel.h: 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/ + +2009-03-18 01:55 +0000 [r182802-182808] Kevin P. Fleming + + * main/astobj2.c, main/asterisk.exports (added), + res/res_odbc.exports (added), res/res_speech.exports (added), + res/res_config_odbc.c, res/res_features.exports (added), + build_tools/strip_nonapi (removed), res/res_adsi.exports (added), + res/res_indications.c, default.exports (added), makeopts.in, + res/res_jabber.exports (added), res/res_monitor.exports (added), + res/res_config_pgsql.c, res/res_snmp.c, main/Makefile, + res/res_smdi.exports (added), include/asterisk/astobj2.h, + res/res_crypto.c, res/res_agi.exports (added), Makefile.rules, + res/res_musiconhold.c: 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). + + * main/astobj2.c, main/asterisk.exports (removed), + res/res_odbc.exports (removed), main/channel.c, + res/res_config_odbc.c, res/res_features.exports (removed), + default.exports (removed), include/asterisk/frame.h, + res/res_jabber.exports (removed), res/res_config_pgsql.c, + main/Makefile, res/res_smdi.exports (removed), + include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c, + res/res_agi.exports (removed), res/res_speech.exports (removed), + include/asterisk/linkedlists.h, main/file.c, + build_tools/strip_nonapi (added), res/res_adsi.exports (removed), + res/res_indications.c, makeopts.in, apps/app_mixmonitor.c, + apps/app_chanspy.c, res/res_monitor.exports (removed), + main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c, + apps/app_meetme.c, res/res_snmp.c, Makefile.rules, + res/res_musiconhold.c: revert commit that included extranous + changes + + * /: remove accidentally merged properties + + * main/astobj2.c, main/asterisk.exports (added), + res/res_odbc.exports (added), main/channel.c, + res/res_config_odbc.c, res/res_features.exports (added), + default.exports (added), include/asterisk/frame.h, + res/res_jabber.exports (added), res/res_config_pgsql.c, + main/Makefile, res/res_smdi.exports (added), + include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c, + res/res_agi.exports (added), res/res_speech.exports (added), + include/asterisk/linkedlists.h, main/file.c, + build_tools/strip_nonapi (removed), res/res_adsi.exports (added), + res/res_indications.c, makeopts.in, apps/app_mixmonitor.c, + apps/app_chanspy.c, res/res_monitor.exports (added), + main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c, + /, apps/app_meetme.c, res/res_snmp.c, Makefile.rules, + res/res_musiconhold.c: 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). + +2009-03-17 20:13 +0000 [r182652] Jason Parker + + * channels/chan_dahdi.c, apps/app_flash.c: Allow dahdichanname to + work as advertised. (closes issue #14056) Reported by: dsedivec + Patches: load_from_zapata_conf.patch uploaded by dsedivec + (license 638) + +2009-03-17 05:50 +0000 [r182449] Tilghman Lesher + + * main/db.c: 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 + +2009-03-16 Leif Madsen + + * Released 1.4.24 + +2009-03-06 Leif Madsen + + * Released 1.4.24-rc1 + +2009-03-06 18:23 +0000 [r180567] Mark Michelson + + * apps/app_voicemail.c: Make compilation succeed in dev-mode when + IMAP storage is enabled. + +2009-03-06 17:19 +0000 [r180532] David Vossel + + * main/enum.c: Fix handling of backreferences for ENUM lookups + enum.c did not handle regex backtraces correctly. The '\1' in the + regex is a backreference that requires a pattern match to be + inserted. The way the code used to work is that it would find the + backreference and insert the entire input string minus the '+'. + This is incorrect. The regexec() function takes in a variable + called pmatch which is an array of structs containing the start + and end indexes for each backreference substring. The original + code actually passed the pmatch array pointer into regexec but + never did anything with it. Now when a backtrace is found, the + backtrace number is looked up in the pmatch array and the correct + substring is inserted. (closes issue #14576) Reported by: + chris-mac Review: http://reviewboard.digium.com/r/187/ + +2009-03-05 23:26 +0000 [r180380-180464] Mark Michelson + + * apps/app_voicemail.c: [IMAP] Fix message retrieval issues when + identical mailbox names were defined in separate contexts. There + was a fix put in a while back so that an X-Asterisk-VM-Context + message header was added to stored IMAP voicemails. This would + allow for us to differentiate if the same mailbox name was used + in multiple contexts. The problem still left was that not all + places where messages were retrieved actually attempted to use + this header for information when retrieving messages. This commit + fixes that so that MWI and message retrieval from VoiceMailMain + work as expected. (closes issue #13853) Reported by: vicks1 + Patches: 13853_v2.patch uploaded by mmichelson (license 60) + Tested by: lmadsen + + * apps/app_voicemail.c, configs/voicemail.conf.sample: Fix broken + mailbox parsing when searchcontexts option is enabled. When using + the searchcontexts option in voicemail.conf, the code made the + assumption that all mailbox names defined were unique across all + contexts. However, the code did nothing to actually enforce this + assumption, nor did it do anything to alert a user that he may + have created an ambiguity in his voicemail.conf file by defining + the same mailbox name in multiple contexts. With this change, we + now will issue a nice long warning if searchcontexts is on and we + encounter the same mailbox name in multiple contexts and ignore + any duplicates after the first box. Whether searchcontexts is + enabled or not, if we come across a duplicate mailbox in the same + context, then we will issue a warning and ignore the duplicated + mailbox. I have also added a small note to voicemail.conf.sample + in the explanation for searchcontexts explaining that you cannot + define the same mailbox in multiple contexts if you have enabled + the option. (closes issue #14599) Reported by: lmadsen Patches: + 14599.patch uploaded by mmichelson (license 60) (with slight + modification) Tested by: lmadsen + +2009-03-05 18:22 +0000 [r180372] Kevin P. Fleming + + * main/rtp.c, main/frame.c, include/asterisk/frame.h: Fix problems + when RTP packet frame size is changed During some code analysis, + I found that calling ast_rtp_codec_setpref() on an ast_rtp + session does not work as expected; it does not adjust the + smoother that may on the RTP session, in fact it summarily drops + it, even if it has data in it, even if the current format's + framing size has not changed. This is not good. This patch + changes this behavior, so that if the packetization size for the + current format changes, any existing smoother is safely updated + to use the new size, and if no smoother was present, one is + created. A new API call for smoothers, + ast_smoother_reconfigure(), was required to implement these + changes. Review: http://reviewboard.digium.com/r/184/ + +2009-03-04 19:22 +0000 [r180194] Joshua Colp + + * main/callerid.c: Look for the number in a callerid string + starting from the end. This way a value using <> can exist in the + name portion. (issue #AST-194) + +2009-03-03 23:01 +0000 [r180010] Jason Parker + + * channels/chan_dahdi.c: Make sure we still support zapchan in + users.conf, in addition to dahdichan. + +2009-03-03 22:48 +0000 [r180006] Mark Michelson + + * configs/queues.conf.sample, apps/app_queue.c: Clarify some + documentation of queues.conf.sample It had always been possible + to explicitly specify a "blank" value for a sound file in + queues.conf and have no sound played back. The problem with this + is that it would result in some ugly CLI warnings from file.c. + This commit introduces a check when playing a file in app_queue + to see if the name of the file is zero-length and return early if + that is the case. Also, the ability to specify the blank sound + files in queues.conf is now mentioned more clearly in + queues.conf.sample (closes issue #14227) Reported by: caspy + +2009-03-03 18:27 +0000 [r179840] Joshua Colp + + * res/res_features.c: Do not assume that the bridge_cdr is still + attached to the channel when the 'h' exten is finished executing. + It is possible for a masquerade operation to occur when the 'h' + exten is operating. This operation moves the CDR records around + causing the bridge_cdr to no longer exist on the channel where it + is expected to. We can not safely modify it afterwards because of + this, so don't even try. (closes issue #14564) Reported by: meric + +2009-03-03 18:11 +0000 [r179807] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, + utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y, + main/ast_expr2f.c: These changes allow AEL to better check ${} + constructs within $[...], that are concatenated with text. I + modified and added rules in ast_expr2.fl to better handle the + concatenations. I added some default routines to ast_expr2.y so + the standalone would compile. It also looks like I haven't run + this thru bison since 2.1, so it's good to get this updated. The + Makefile has comments added now for check_expr2 and check_expr to + explain what they are for, and how to run them. The testexpr2s + stuff has been removed, in favor of check_expr2. expr2.testinput + has been updated to include the two expressions that inspired + these changes (from mcnobody on #asterisk this morning) The + regression has been run and all looks well. + +2009-03-03 16:45 +0000 [r179741] Russell Bryant + + * main/channel.c: Ensure chan->fdno always gets reset to -1 after + handling a channel fd event. Since setting fdno to -1 had to be + moved, a couple of other code paths that do process an fd event + return early and do not pass through the code path where it was + moved to. So, set it to -1 in a few other places, too. + +2009-03-03 14:38 +0000 [r179671] Joshua Colp + + * main/channel.c: Move where fdno is set to the default value to + *after* the read callback of the channel driver is called. We + have to do this as the underlying channel driver may need the + fdno value to determine what to read. + +2009-03-03 13:53 +0000 [r179608] Russell Bryant + + * main/channel.c: Make it easier to detect an improper call to + ast_read(). When you call ast_waitfor() on a channel, the index + into the channel fds array that holds the file descriptor that + poll() determines has input available is stored in fdno. This + patch clears out this value after a call to ast_read() and also + reports errors if ast_read() is called without an fdno set. From + a discussion on the asterisk-dev list. + +2009-03-02 23:54 +0000 [r179536] Jeff Peeler + + * main/channel.c: Fix bridging regression from commit 176701 This + fixes a bad regression where the bridge would exit after an + attended transfer was made. The problem was due to nexteventts + getting set after the masquerade which caused the bridge to + return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: + tim_ringenbach + +2009-03-02 23:34 +0000 [r179532] Russell Bryant + + * apps/app_meetme.c: Move ast_waitfor() down to avoid the results + of the API call becoming stale. This call to ast_waitfor() was + being done way too soon in this section of code. Specifically, + there was code in between the call to waitfor and the code that + uses the result that puts the channel in autoservice. By putting + the channel in autoservice, the previous results of ast_waitfor() + become meaningless, as the autoservice thread will do it's own + ast_waitfor() and ast_read() on the channel. So, when we came + back out of autoservice and eventually hit the block of code that + calls ast_read() on the channel, there may not actually be any + input on the channel available. Even though the previous call to + ast_waitfor() in app_meetme said there was input, the autoservice + thread has since serviced the channel for some period of time. + This bug manifested itself while dvossel was doing some testing + of MeetMe in Asterisk trunk. He was using the timerfd timing + module. When the code hit ast_read() erroneously, it determined + that it must have been called because of input on the timer fd, + as chan->fdno was set to AST_TIMING_FD, since that was the cause + of the last legitimate call to ast_read() done by autoservice. In + this test, an IAX2 channel was calling into the MeetMe + conference. It was _much_ more likely to be seen with an IAX2 + channel because of the way audio is handled. Every audio frame + that comes in results in a call to ast_queue_frame(), which then + uses ast_timer_enable_continuous() to notify the channel thread + that a frame is waiting to be handled. So, the chances of + ast_waitfor() indicating that a channel needs servicing due to a + timer event on an IAX2 event is very high. Finally, it is + interesting to note that if a different timing interface was + being used, this bug would probably not be noticed. When + ast_read() is called and erroneously thinks that there is a timer + event to handle, it calls the ast_timer_ack() function. The + pthread and dahdi timing modules handle the ack() function being + called when there is no event by simply ignoring it. In the case + of the timerfd module, it results in a read() on the timer fd + that will block forever, as there is no data to read. This caused + Asterisk to lock up very quickly. Thanks to dvossel and + mmichelson for the fun debugging session. :-) + +2009-03-02 23:09 +0000 [r179468] Tilghman Lesher + + * main/app.c: When ending a recording with silence detection, + remember to reduce the duration. The end of the recording is + correspondingly trimmed, but the duration was not trimmed by the + number of seconds trimmed, so the saved duration was necessarily + longer than the actual soundfile duration. (closes issue #14406) + Reported by: sasargen Patches: 20090226__bug14406.diff.txt + uploaded by tilghman (license 14) Tested by: sasargen + +2009-03-02 22:58 +0000 [r179461] Russell Bryant + + * main/channel.c: Ensure that only one thread is calling + ast_settimeout() on a channel at a time. For example, with an + IAX2 channel, you can have both the channel thread and the + chan_iax2 processing threads calling this function, and doing so + twice at the same time is a bad thing. (Found in a debugging + session with dvossel and mmichelson) + +2009-03-02 20:14 +0000 [r179395] Jason Parker + + * main/editline/configure, main/editline/np/unvis.c, + main/editline/sys.h, main/editline/configure.in: Remove several + silly warnings in editline. One about a broken preprocessor + directive, and another about strlcpy/strlcat. (closes issue + #14264) Reported by: dimas + +2009-02-27 19:03 +0000 [r179056] Jason Parker + + * doc/channelvariables.txt: Update documentation for DIALEDTIME and + ANSWEREDTIME variables. (closes issue #14566) Reported by: + klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by + klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by + klaus3000 (license 65) + +2009-02-26 21:27 +0000 [r178956] Steve Murphy + + * configs/features.conf.sample, res/res_features.c: This change + moves the default feature digit timeout to 1000 ms from the + previous default of 500. As per bug 14515, a dev discussion + arrived at a "mediated concensus" of a default feature digit + timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for + distracted phone users in phone booths; kpfleming put his foot + down at 1.0 sec. Users who found the previous default max delay + of 250 msec perfect, are welcome to override the new default. + Notice that I said that 250 msec was the default; wait a minute, + you might say, the config file said it was 500 msec!; well, + because of the bug fix for 14515, we found that 500 msec was + actually enforcing a max of 250. The bug fix would restore 500 + msec, but we felt even that was a bit tight for most users... + 2000 msec was pushed earlier by mmichelson, so that reduces to + 1000 msec after the bug fix. Enjoy! + +2009-02-26 17:24 +0000 [r178838] David Vossel + + * channels/chan_iax2.c: IAX2 prune realtime fix Now prune_users() + and prune_peers() are called instead of reload_config() to prune + all users/peers that are realtime. These functions remove all + users/peers with the rtfriend and delme flags set. + iax2_prune_realtime() also lacked the code to properly delete a + single friend. For example. if iax2 prune realtime was + called, only the peer instance would be removed. The user would + still remain. (closes issue #14479) Reported by: mousepad99 + Review: http://reviewboard.digium.com/r/176/ + +2009-02-26 17:09 +0000 [r178640-178804] Steve Murphy + + * res/res_features.c: This patch prevents the feature detection + timeout from being cut in half. Because the ast_channel_bridge() + call will return 0 and pass a frame pointer for both DTMF_BEGIN + and DTMF_END, the feature_timer field in hte config struct is + getting decremented twice, which effectively cuts the + digittimeout in half. I added conditions to the if statement to + only let DTMF_END frames to flow thru, which solved the problem. + Also, when the frame pointer is null, let control flow thru-- + this usually happens on timeouts. I added a comment to the code + to explain what's going on and why. Many thanks to sodom for + reporting this problem. Personnally, it always seemed like + something was wrong with the featuredigittimeout, but I never + could quite decide what... and was too busy to investigate. This + bug forced the issue, and now we know. Sodom had other issues in + 14515, but I couldn't reproduce them. If he still has problems, + and wants to get them solved, he is welcome to reopen 14515. + (closes issue #14515) Reported by: sodom Patches: 14515.patch + uploaded by murf (license 17) Tested by: murf, sodom + + * main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the + fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) + 8-bit transparent While I was updating ast_expr2.fl, I missed one + rule that would allow 8-bit chars to be caught in tokens; and in + so doing, it absorbs the ${ sequence and messes up the checking + of raw exprs by AEL. Trunk already has these changes. (closes + issue #14543) Reported by: klaus3000 Patches: patch.14543 + uploaded by murf (license 17) Tested by: murf + +2009-02-25 12:43 +0000 [r178508] Russell Bryant + + * main/asterisk.c: Update the copyright year for the main page of + the doxygen documentation. + +2009-02-24 23:25 +0000 [r178445] Tilghman Lesher + + * configs/extensions.conf.sample: Add section about the #exec + command in configuration files. (closes issue #14540) Reported + by: jtodd Patch by: jtodd, with additional notes by tilghman + (license 14) + +2009-02-24 20:36 +0000 [r178373] Russell Bryant + + * main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset + to 0 properly. (issue #14460) Reported by: moliveras Tested by: + russell + +2009-02-24 17:02 +0000 [r178266] Terry Wilson + + * apps/app_dahdiras.c, res/res_musiconhold.c: Change include order + to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets + defined before linux/types.h is included, the __s32 type doesn't + get defined + +2009-02-24 15:16 +0000 [r178205] Joshua Colp + + * channels/chan_sip.c: Skip check for extension when subscribing + for MWI. Since the remote side is not actually subscribing to a + specific extension when subscribing for MWI just skip the check + to see if the extension exists. They can't use it to specify the + mailbox either since we require configuration of that in sip.conf + (closes issue #14531) Reported by: festr + +2009-02-23 23:09 +0000 [r178141] Russell Bryant + + * main/rtp.c: Fix infinite DTMF when a BEGIN is received without an + END. This commit is related to rev 175124 of 1.4 where a previous + attempt was made to fix this problem. The problem with the + previous patch was that the inserted code needed to go _before_ + setting the lastrxts to the current timestamp. Because those were + the same, the dtmfcount variable was never decremented, and so + the END was never sent. In passing, I removed the dtmfsamples + variable which was completed unused. I also removed a redundant + setting of the lastrxts variable. (closes issue #14460) Reported + by: moliveras + +2009-02-20 22:59 +0000 [r177701-177786] Tilghman Lesher + + * main/pbx.c: Don't print the CR-NL combination when we aren't + outputting to the manager. An embedded CR-NL in a CLI command + screws up several AMI parsers that don't expect to see that + combination in the middle of output. (Closes issue #14305) + Reported by: martins Patch by: tilghman + + * include/asterisk/threadstorage.h: This exception does not appear + to still be true for Solaris 10, and OpenSolaris definitely needs + it to be removed. Fixed for snuff-home on -dev channel. + +2009-02-20 20:17 +0000 [r177696] David Vossel + + * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with + undefined audio codecs in chan_iax2 During iax2 call negotiation, + supported codecs are passed in an Information Element containing + a 2 byte field where each bit correlates to a specific codec. In + 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 + undefined. By default all bits are enabled unless specified + otherwise. Since its a 2 byte field and 13-15 are not defined, + these bits are never turned off. In trunk, bits 13-15 are + defined, which means 1.4 is advertising support for codecs it + does not have when talking to trunk. I fixed this by adding + #define for undefined audio codec bits. These bits are then + removed from iax2's full bandwidth capabilities. (closes issue + #14283) Reported by: jcovert + +2009-02-19 22:51 +0000 [r177540] Steve Murphy + + * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch + fixes a problem with 8-bit input to the ast_expr2 scanner. The + real culprit was the --full argument to flex in the Makefile! + This causes a 7-bit scanner to be generated. I reviewed the rules + and found one rule where I needed to specifically include 8-bit + chars for a token. I tested against the text supplied by ibercom, + and all looks very well. This has been there a surprisingly long + time! (closes issue #14498) Reported by: ibercom Patches: + 14498.patch uploaded by murf (license 17) Tested by: murf + +2009-02-19 22:26 +0000 [r177536] Tilghman Lesher + + * apps/app_voicemail.c: Fix up potential crashes, by reducing the + sharing between interactive and non-interactive threads. (closes + issue #14253) Reported by: Skavin Patches: + 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) + Tested by: Skavin + +2009-02-19 18:58 +0000 [r177450] Olle Johansson + + * channels/chan_sip.c: Force a MWI notification after subscribe + request. Reported by the Resiprocate dev team. Thanks! + +2009-02-19 16:37 +0000 [r177383] Joshua Colp + + * apps/app_speech_utils.c: If we are able to create a speech + structure unset the ERROR variable in case it was previously set. + (issue #LUMENVOX-13) + +2009-02-18 22:43 +0000 [r177225] Steve Murphy + + * pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression + of sorts that was introduced in rev 24425. It basically fixes + AST-190/ABE-1782. What was wrong: the user has 6000 extensions in + one context; and then 6000 contexts, one per extension. The + parser could only handle about 4893 of the 6000 extens in the + single context. This was due to the regression I mentioned. To + get rid of shift/reduce conflicts, Luigi set up right-recursive + lists for globals, context elements, switch lists, and + statements. Right recursive lists got rid of the warnings, but + instead, they use up a tremendous amount of stack space when the + lists are long. I saw this a few years back, and resolved not to + fix it until someone complained. That day has arrived! After the + changes were made, I ran the regression test suite, and there + were no problems. I took the test case the user provided, and + added 100,000 extensions to the single context, that already had + 6,000 extens in it. (I'll see your 6, and raise you 100!) It + takes a few minutes to read it all in, check it and generate code + for it, but no problems. So, I think I can say that + fundamentally, there are no longer any limits on the number of + items you can place in contexts, statement blocks, switches, or + globals, beyond your virt mem constraints. + +2009-02-18 20:06 +0000 [r177160] Jeff Peeler + + * channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx, + autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h, + channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, + channels/h323/ast_ptlib.h (added), configure, + channels/h323/compat_h323.h, configure.ac, + channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323 + to build against PTLib as well as the older PWLib Several changes + in PTLib have occurred requiring build time detection. Changes + accounted for include the library name change, config option + change, install location change, and a boolean type change which + is handled by ast_ptlib.h. Also, the sed check has been modified + to properly work with autoconf >= 2.62. (closes issue #14224) + Reported by: bergolth Patches: asterisk-autoconf-sed.patch + uploaded by bergolth (license 661) asterisk-pwlib-v3.patch + uploaded by bergolth (license 661) Tested by: jpeeler + +2009-02-18 18:30 +0000 [r177096] Tilghman Lesher + + * include/asterisk/config.h: Document the return value of the + update method (as requested on -dev list) + +2009-02-18 17:41 +0000 [r176945-177039] Doug Bailey + + * main/utils.c: Merged revisions 177035 manually from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | + dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines + Fixed error where a check for an zero length, terminated string + was needed. ........ + + * main/utils.c: Need to take into account the \0 terminator of the + old string to determine the amount available. + +2009-02-18 00:34 +0000 [r176810] Shaun Ruffell + + * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice + with G723. This commit brings in the changes that were living out + on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch. + codec_dahdi.c now always uses signed linear as the simple codec + so that a soft g729 codec will not end up being preferred to the + hardware codec. There are also changes to allow codec_dahdi.c to + feed packets to the hardware in the native sample size of the + codec. This solves problems with choppy audio when using G723. + +2009-02-17 21:54 +0000 [r176701] Jeff Peeler + + * main/channel.c, res/res_features.c, include/asterisk/channel.h: + Modify bridging to properly evaluate DTMF after first warning is + played The main problem is currently if the Dial flag L is used + with a warning sound, DTMF is not evaluated after the first + warning sound. To fix this, a flag has been added in + ast_generic_bridge for playing the warning which ensures that if + a scheduled warning is missed, multiple warrnings are not played + back (due to a feature evaluation or waiting for digits). + ast_channel_bridge was modified to store the nexteventts in the + ast_bridge_config structure as that information was lost every + time ast_channel_bridge was reentered, causing a hangup due to + incorrect time calculations. (closes issue #14315) Reported by: + tim_ringenbach Reviewed on reviewboard: + http://reviewboard.digium.com/r/163/ + +2009-02-17 21:21 +0000 [r176426-176661] Tilghman Lesher + + * channels/chan_local.c: Backport change to 1.4: Prior to + masquerade, move the group definitions to the channel performing + the masq, so that the group count lingers past the bridge. + (closes issue #14275) Reported by: kowalma Patches: + 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) + Tested by: kowalma + + * channels/chan_sip.c: After a 'sip reload', qualifies for realtime + peers weren't immediately restarted, instead waiting until the + next registration. We're now caching the qualify across a + reload/restart and starting the qualify immediately upon loading + the peer. (closes issue #14196) Reported by: pdf Patches: + 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) + Tested by: pdf + +2009-02-16 23:30 +0000 [r176354] David Vossel + + * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not + being relayed correctly during bridging This should have been + committed with rev176247, but I missed it. srcupdate frames no + longer break out of the native bridge, but are not being sent to + the other call leg either. This fixs that. issue #13749 + +2009-02-16 21:41 +0000 [r176254] Kevin P. Fleming + + * main/utils.c: correct a logic error in the last stringfields + commit... don't mark additional space as allocated if the string + was built using already-allocated space + +2009-02-16 21:39 +0000 [r176249-176252] Mark Michelson + + * apps/app_meetme.c: Remove unused variable and make dev-mode + compilation happy + + * apps/app_meetme.c: Open the DAHDI pseudo device and set it to be + nonblocking atomically Apparently on FreeBSD, attempting to set + the O_NONBLOCKING flag separately from opening the file was + causing an "inappropriate ioctl for device" error. While I cannot + fathom why this would be happening, I certainly am not opposed to + making the code a bit more compact/efficient if it also fixes a + bug. (closes issue #14482) Reported by: ys Patches: meetme.patch + uploaded by ys (license 281) Tested by: ys + +2009-02-16 21:28 +0000 [r176247] David Vossel + + * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE + breaking out of native bridge In iax2, when a + AST_CONTROL_SRCUPDATE is received it breaks from the native + bridge, but since there is no code path to handle srcupdate it + just goes to be beginning of the loop. This was causing packet + storms of srcupdate frames between servers. Now srcupdate frames + do not break the native bridge for processing. (closes issue + #13749) Reported by: adiemus + +2009-02-16 21:10 +0000 [r176216] Kevin P. Fleming + + * main/utils.c: fix a flaw in the ast_string_field_build() family + of API calls; these functions made no attempt to reuse the space + already allocated to a field, so every time the field was written + it would allocate new space, leading to what appeared to be a + memory leak. + +2009-02-16 15:33 +0000 [r176029] Joshua Colp + + * channels/chan_sip.c: Don't have the Via header stored as a + stringfield as it can change often during the lifetime of a + dialog. This issue crept up with subscriptions on the AA50. When + an outgoing NOTIFY is sent a new branch value is created and the + Via header is changed to reflect it. Since this was a stringfield + a new spot in the pool was used for the value while the old was + left untouched/unused. If the current pool was full a new pool + was created. This would cause memory usage to increase steadily. + (issue #AA50-2332) + +2009-02-15 23:37 +0000 [r175921] Michiel van Baak + + * main/pbx.c, channels/chan_sip.c, main/devicestate.c, + include/asterisk/manager.h: fix mis-spelling of the word + registered. Reported by De_Mon on #asterisk-dev. + +2009-02-15 20:33 +0000 [r175777-175825] Olle Johansson + + * formats/format_ilbc.c: format_ilbc does not depend on codec + libraries and can therefore always be made. My mistake. Ursäkta! + + * formats/format_ilbc.c: Disable format_ilbc.so by default, like + codec_ilbc.so + + * channels/chan_sip.c: Make sure that the debug line is not printed + on debug level 0 + +2009-02-13 21:53 +0000 [r175698] Jason Parker + + * include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather, + Zaptel is actually Zaptel. (in case you're confused, DAHDI is + still DAHDI) + +2009-02-13 19:47 +0000 [r175407-175590] Mark Michelson + + * apps/app_voicemail.c: Fix a potential crash situation when using + IMAP voicemail If calling into VoiceMailMain when using IMAP + storage, it was possible to crash Asterisk by hanging up the + phone when prompted for a voicemail mailbox. This patch fixes the + issue. While it may appear that this patch is superficial, it + allows code execution to continue to the failure case just below + the IMAP_STORAGE code block where this patch has been applied + (closes issue #14473) Reported by: dwpaul Patches: + voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license + 689) + + * main/file.c: Fix a place where filestreams were not refcounted + properly This section was already present in trunk and other + branches, but did not exist in 1.4. (closes issue #14395) + Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut + (license 60) Tested by: ZX81 + +2009-02-12 21:19 +0000 [r175311] Tilghman Lesher + + * main/udptl.c: Fix crashes when receiving certain T.38 packets. + Also, increase the maximum size of T.38 packets and warn users + when they try to set the limits above those maximums. (closes + issue #13050) Reported by: schern Patches: + 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) + Tested by: schern + +2009-02-12 20:34 +0000 [r175187-175294] Jeff Peeler + + * res/res_features.c: Fix ParkedCall event information for From + field in the case of a blind transfer If the parker information + can not be obtained from the peer, try and see if the + BLINDTRANSFER channel variable has been set. Previously, a blind + transfer to the ParkAndAnnounce app would return nothing for the + From. Closes AST-189 + + * res/res_features.c: Fix crash in event of failed attempt to + transfer to parking The peer may not necessarily exist, such as + in the case of a transfer to ParkAndAnnounce. In this case don't + try to play a sound to it. + +2009-02-12 16:51 +0000 [r175124] Russell Bryant + + * main/rtp.c: Don't send DTMF for infinite time if we do not + receive an END event. I thought that this was going to end up + being a pretty gnarly fix, but it turns out that there was + actually already a configuration option in rtp.conf, dtmftimeout, + that was intended to handle this situation. However, in between + Asterisk 1.2 and Asterisk 1.4, the code that processed the option + got lost. So, this commit brings it back to life. The default + timeout is 3 seconds. However, it is worth noting that having + this be configurable at all is not really the recommended + behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the + time period of extending the tone is necessary to avoid that a + tone "gets stuck". Regardless of the algorithm used, the tone + SHOULD NOT be extended by more than three packet interarrival + times. A slight extension of tone durations and shortening of + pauses is generally harmless. Three seconds will pretty much + _always_ be far more than three packet interarrival times. + However, that behavior is not required, so I'm going to leave it + with our legacy behavior for now. Code from + svn/asterisk/team/russell/issue_14460 (closes issue #14460) + Reported by: moliveras + +2009-02-12 10:16 +0000 [r175029] Philippe Sultan + + * channels/chan_gtalk.c: Set the initiator attribute to lowercase + in our replies when receiving calls. This attribute contains a + JID that identifies the initiator of the GoogleTalk voice + session. The GoogleTalk client discards Asterisk's replies if the + initiator attribute contains uppercase characters. (closes issue + #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded + by jcovert (license 551) Tested by: jcovert + +2009-02-12 00:19 +0000 [r174997] Joshua Colp + + * main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy + commit by russell via SMS. + +2009-02-12 00:01 +0000 [r174985-174986] Russell Bryant + + * main/rtp.c: Clear out the current event after forcing the end of + a digit + + * main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END + event is ever received + +2009-02-11 20:54 +0000 [r174885] Tilghman Lesher + + * main/pbx.c, apps/app_macro.c: Restore a behavior that was + recently changed, when we fixed issue #13962 and issue #13363 + (related to issue #6176). When a hangup occurs during a Macro + execution in earlier 1.4, the h extension would execute within + the Macro context, whereas it was always supposed to execute only + within the main context (where Macro was called). So this fix + checks for an "h" extension in the deepest macro context where a + hangup occurred; if it exists, that "h" extension executes, + otherwise the main context "h" is executed. (closes issue #14122) + Reported by: wetwired Patches: 20090210__bug14122.diff.txt + uploaded by Corydon76 (license 14) Tested by: andrew + +2009-02-10 18:50 +0000 [r174644] Joshua Colp + + * channels/chan_sip.c: Go off hold when we get an empty reinvite + telling us to. (closes issue #14448) Reported by: frawd Patches: + hold_invite_nosdp.patch uploaded by frawd (license 610) + +2009-02-10 17:52 +0000 [r174583] Matthew Nicholson + + * main/jitterbuf.c: Improve behavior of jitterbuffer when + maxjitterbuffer is set. This change improves the way the + jitterbuffer handles maxjitterbuffer and dramatically reduces the + number of frames dropped when maxjitterbuffer is exceeded. In the + previous jitterbuffer, when maxjitterbuffer was exceeded, all new + frames were dropped until the jitterbuffer is empty. This change + modifies the code to only drop frames until maxjitterbuffer is no + longer exceeded. Also, previously when maxjitterbuffer was + exceeded, dropped frames were not tracked causing stats for + dropped frames to be incorrect, this change also addresses that + problem. (closes issue #14044) Patches: bug14044-1.diff uploaded + by mnicholson (license 96) Tested by: mnicholson Review: + http://reviewboard.digium.com/r/144/ + +2009-02-10 02:27 +0000 [r174369] Steve Murphy + + * apps/app_rpt.c: This patch solves some compiler complaints in + both 32 and 64-bit environments. + +2009-02-09 17:11 +0000 [r174282] Mark Michelson + + * channels/chan_sip.c: Don't do an SRV lookup if a port is + specified RFC 3263 says to do A record lookups on a hostname if a + port has been specified, so that's what we're going to do. See + section 4.2. (closes issue #14419) Reported by: klaus3000 + Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by + klaus3000 (license 65) + +2009-02-09 14:48 +0000 [r174218] Joshua Colp + + * res/res_musiconhold.c: Don't overwrite our pointer to the music + class when music on hold stops. We will use this if it starts + again to see if we can resume the music where it left off. + (closes issue #14407) Reported by: mostyn + +2009-02-07 16:15 +0000 [r174148] Russell Bryant + + * res/snmp/agent.c: Fix a race condition that could cause a crash. + +2009-02-06 23:36 +0000 [r174082] Dwayne M. Hubbard + + * channels/chan_sip.c: check ast_strlen_zero() before calling + ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() + The reporter didn't actually upload a properly-formed patch, + instead a modified chan_sip.c file was uploaded. I created a + patch to determine the changes, then modified the suggested + changes to create a proper fix. The summary above is a complete + description of the changes. (closes issue #13547) Reported by: + tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license + 258) Tested by: tecnoxarxa + +2009-02-06 17:15 +0000 [r173967-173968] Joshua Colp + + * channels/chan_sip.c: Remove a debug message I put in by accident. + + * channels/chan_sip.c: Some clients do not put the call-id for + replaces at the beginning, so support it being anywhere in the + string. (closes issue #14350) Reported by: fhackenberger + +2009-02-06 16:20 +0000 [r173917] Matthew Nicholson + + * channels/chan_sip.c: Limit the addition of the Contact header in + SIP responses according to various SIP RFCs. (closes issue + #13602) Reported by: hjourdain Tested by: mnicholson + +2009-02-06 15:43 +0000 [r173900] Tilghman Lesher + + * utils/muted.c: Backport OS X fix from trunk (AGAIN, closes issue + #14360) + +2009-02-05 23:19 +0000 [r173770] Mark Michelson + + * channels/chan_sip.c: Fix logic regarding when to perform an SRV + lookup for outgoing REGISTER requests With this fix, we only will + perform an SRV lookup at the following times: * The first time we + register with a remote registrar * If we send a REGISTER but do + not receive a response * If the sendto() function returns an + error While I wrote the patch that fixes this issue, a huge + amount of credit is due to Brett Bryant, who wrote the initial + patch on which I based this one. (closes issue #12312) Reported + by: jrast Patches: 12312.patch uploaded by putnopvut (license 60) + Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/ + +2009-02-05 20:47 +0000 [r173696] Jeff Peeler + + * apps/app_voicemail.c: Add new configuration option to make shared + IMAP mailboxes function as expected. The new option is + "imapvmshareid" which is an ID to tag multiple mailboxes using + the same IMAP storage location to function as one mailbox. This + allows all messages to be retrieved for any user in the group. + The patch alters the 'X-Asterisk-VM-Extension' header that is + responsible for matching voicemails for a given user. (closes + issue #13673) Reported by: howardwilkinson + +2009-02-05 20:29 +0000 [r173392-173692] Mark Michelson + + * apps/app_queue.c: Fix situations where queue members could be + autopaused unexpectedly Specifically, this patch prevents us from + autopausing members when we receive a busy or congestion frame + from them. (closes issue #14376) Reported by: fiddur Patches: + 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur + + * apps/app_mixmonitor.c: Add some missing cleanup to app_mixmonitor + + * apps/app_mixmonitor.c: Fix a problem where a channel pointer + becomes invalid due to masquerading or hanging up. app_mixmonitor + runs its own thread to monitor the channel's activity and write + the mixed audio to a file. Since this thread runs independently + of the channel, it is possible that the mixmonitor thread's + channel pointer will point to freed memory when the channel + either is masqueraded or hangs up (technically, both cases are + hangups, but we need to handle the cases slightly differently). + The solution for this is to employ a datastore, which has the + nice benefit of allowing us to hook into channel masquerades and + hangups and update our pointer as necessary. If this looks + familiar, this same technique is employed in app_chanspy. + app_chanspy is a bit more involved since it does a lot more + operations on the channel that is being spied upon. + app_mixmonitor does have an extra touch that app_chanspy doesn't + have, though. Since there is a thread race between the channel's + thread and the mixmonitor thread on a hangup, we em- ploy a + condition-and-boolean combination to ensure that the channel + thread finishes with our structure before the mixmonitor thread + attempts to free it. No crashes! (closes issue #14374) Reported + by: aragon Patches: 14374.patch uploaded by putnopvut (license + 60) Tested by: aragon, putnopvut + + * apps/app_chanspy.c: Revert my previous change because it was + stupid + + * apps/app_chanspy.c: Add a missing unlock. Extremely unlikely to + ever matter, but it's needed. + +2009-02-03 23:35 +0000 [r173248] David Vossel + + * channels/chan_iax2.c: 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/ + +2009-02-03 21:57 +0000 [r173211] Jeff Peeler + + * res/res_features.c: Parking attempts made to one end of a bridge + no longer will hang up due to a parking failure. Parking attempts + made using either one-touch, or doing either a blind or assisted + transfer to the parking extension now keep up the bridge instead + of hanging up the attempted parked party. Normal causes for the + parking attempt to fail includes the specific specified extension + (via PARKINGEXTEN) not being available or if all the parking + spaces are currently in use. To avoid having to reverse a + masquerade park_space_reserve was made to provide foresight if a + parking attempt will succeed and if so reserve the parking space. + (closes issue #13494) Reported by: mdu113 Reviewed by Russell: + http://reviewboard.digium.com/r/133/ + +2009-02-03 00:15 +0000 [r173070] Tilghman Lesher + + * configs/extensions.conf.sample: Add warning to standard config, + that globals may be overridden by other dialplan configuration + files. (closes issue #14388) Reported by: macli + +2009-02-02 23:48 +0000 [r173066] Terry Wilson + + * res/res_features.c: Fix a feature inheritance bug I added after + code review + +2009-02-02 20:28 +0000 [r172962] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: + 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 + +2009-01-31 00:15 +0000 [r172517-172639] Terry Wilson + + * configs/features.conf.sample, res/res_features.c: 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 + + * configs/features.conf.sample, apps/app_dial.c, + main/global_datastores.c, res/res_features.c, + doc/channelvariables.txt, include/asterisk/global_datastores.h, + CHANGES: 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/ + +2009-01-29 22:54 +0000 [r172438] Tilghman Lesher + + * main/asterisk.c, apps/app_nbscat.c, autoconf/ast_func_fork.m4, + apps/app_festival.c, build_tools/menuselect-deps.in, configure, + apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c, + apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: + 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 + +2009-01-29 08:48 +0000 [r172169] Olle Johansson + + * channels/chan_sip.c: 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 + +2009-01-28 18:51 +0000 [r172030] Steve Murphy + + * apps/app_channelredirect.c, main/pbx.c, main/manager.c, + res/res_features.c, include/asterisk/channel.h: 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 + +2009-01-28 17:25 +0000 [r171963] Tilghman Lesher + + * channels/chan_dahdi.c: Clarify log message (suggested by + manxpower on #asterisk-dev) + +2009-01-28 13:07 +0000 [r171837] Olle Johansson + + * configs/sip.conf.sample: Add a better explanation of the + difference between the device namespace and the dialplan for + newbies. + +2009-01-27 21:55 +0000 [r171621-171689] Mark Michelson + + * channels/chan_agent.c: 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 + + * main/slinfactory.c: 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) + +2009-01-27 14:33 +0000 [r171527] Olle Johansson + + * channels/chan_sip.c: Use the same branch tag in CANCEL as in + INVITE Originally putnopvut implemented some changes in revision + 142079 that according to the bug report seemed to have worked + then, but somehow fails now. I guess code, as humans, get old and + forget stuff. Anyway, this bug caused CANCEL not to work with + picky systems. Thanks Fredrik for pointing out where the bug in + the SIP messaging was. (closes issue #14346) Reported by: oej + Patches: bug14346.diff uploaded by oej (license 306) Tested by: + oej + +2009-01-26 21:31 +0000 [r171452] Russell Bryant + + * channels/chan_iax2.c: Resolve some synchronization issues in + chan_iax2 scheduler handling. The important changes here are + related to the synchronization between threads adding items into + the scheduler and the scheduler handling thread. By adjusting the + lock and condition handling, we ensure that the scheduler thread + sleeps no longer and no less than it is supposed to. We also + ensure that it does not wake up more often than it has to. There + is no bug report associated with this. It is just something that + I found while putting scheduler thread handling into a reusable + form (review 129). Review: http://reviewboard.digium.com/r/131/ + +2009-01-26 12:51 +0000 [r171264] Olle Johansson + + * channels/chan_sip.c: Don't retransmit 401 on REGISTER requests + when alwaysauthreject=yes (closes issue #14284) Reported by: + klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt + uploaded by klaus3000 (license 65) Tested by: klaus3000 + +2009-01-25 23:44 +0000 [r171120-171187] Tilghman Lesher + + * channels/chan_oss.c: Correctly track the hookstate (closes issue + #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt + uploaded by Corydon76 (license 14) + + * res/res_agi.c: Err, yeah. + + * res/res_agi.c: Add thread to kill zombies, when child processes + don't die immediately on SIGHUP. (closes issue #13968) Reported + by: eldadran Patches: 20090114__bug13968.diff.txt uploaded by + Corydon76 (license 14) Tested by: eldadran + +2009-01-25 13:33 +0000 [r170979] Sean Bright + + * apps/app_page.c: Resolve a logic error that was causing Page() to + crash when more than one channel was specified. (closes issue + #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt + uploaded by seanbright (license 71) Tested by: kc0bvu + +2009-01-24 13:55 +0000 [r170836] Tilghman Lesher + + * configs/res_odbc.conf.sample: Remove superfluous implementation + note (closes issue #14319) + +2009-01-23 20:55 +0000 [r170671-170719] Mark Michelson + + * configs/res_odbc.conf.sample: Add notes to the idlecheck + explanation in res_odbc.conf.sample (closes issue #14319) + Reported by: klaus3000 Patches: + patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 + (license 65) + + * contrib/i18n.testsuite.conf: Update contrib/i18n.testsuite.conf + to not use deprecated syntax * Convert Wait,1 to Wait(1) * + Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all + priorities beyond the first Also added test for Chinese numbers, + too. (closes issue #14320) Reported by: dant Patches: + i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license + 670) + +2009-01-23 20:16 +0000 [r170648] Joshua Colp + + * main/channel.c: When a channel is answered make sure any + indications currently playing stop. Usually the phone would do + this but if the channel was already answered then they are being + generated by Asterisk and we darn well need to stop them. (closes + issue #14249) Reported by: RadicAlish + +2009-01-23 Tilghman Lesher + + * Asterisk 1.4.23.1 released. + + * channels/chan_iax2.c: Regression fix for AST-2009-001 security + fix. + +2009-01-21 Leif Madsen + + * Asterisk 1.4.23 released. + +2009-01-20 18:49 -0500 [r169581] Terry Wilson + + * One-touch parking was calling back the wrong channel on timeout + +2009-01-20 13:40 -0500 [r169485] Terry Wilson + + * Don't play audio to the channel if we've masqueraded (closes + issue #14066) Reported by: bluefox Tested by: otherwiseguy, + bluefox + +2009-01-16 Russell Bryant + + * Asterisk 1.4.23-rc4 released. + +2009-01-16 00:19 +0000 [r168745] Steve Murphy + + * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump, + in this case) fails a consistency check because it can't find a + matching extension. The problem was a missing instruction to end + the range notation in the code where it converts the pattern into + a regex and uses the regex code to determine the match. I tested + using the AEL code the user supplied, and now, the consistency + check passes. (closes issue #14141) Reported by: dimas + +2009-01-15 18:43 +0000 [r168721] Olle Johansson + + * configs/extconfig.conf.sample: Meetme actually has realtime but + wasn't documented + +2009-01-15 18:22 +0000 [r168716] Terry Wilson + + * res/res_features.c: Convert call to park_call_full to + masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE + return value, we need to use masqueraded parking, otherwise we + will try to call ast_hangup() in __pbx_run() and in + do_parking_thread() and then promptly crash. (closes issue + #14215) Reported by: waverly360 Tested by: otherwiseguy (closes + issue #14228) Reported by: kobaz Tested by: otherwiseguy + +2009-01-15 01:20 +0000 [r168633] Tilghman Lesher + + * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on + security fix AST-2009-001 (Closes issue #14238) + +2009-01-15 00:11 +0000 [r168628] Mark Michelson + + * apps/app_queue.c: Fix some crashes from bad datastore handling in + app_queue.c * The queue_transfer_fixup function was searching for + and removing the datastore from the incorrect channel, so this + was fixed. * Most datastore operations regarding the + queue_transfer datastore were being done without the channel + locked, so proper channel locking was added, too. (closes issue + #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by + putnopvut (license 60) Tested by: ZX81, festr + +2009-01-14 21:48 +0000 [r168622] Richard Mudgett + + * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of + process ID values. The allocated process IDs could overflow their + respective NT and TE fields. Affects outgoing calls. + +2009-01-14 20:52 +0000 [r168614] Sean Bright + + * contrib/scripts/autosupport: Update autosupport script to supply + info for both Zaptel and DAHDI in 1.4 and be sure to run + dahdi_test in 1.6.x and trunk instead of zttest. (closes issue + #14132) Reported by: dsedivec Patches: + asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638) + asterisk-trunk-autosupport.patch uploaded by dsedivec (license + 638) + +2009-01-14 19:34 +0000 [r168608] Steve Murphy + + * apps/app_page.c: app_page was failing to compile in dev-mode on + my gcc-4.2.4 system. This change gets rid of the warning. + +2009-01-14 19:02 +0000 [r168603] Tilghman Lesher + + * main/udptl.c: Don't read into a buffer without first checking if + a value is beyond the end. (closes issue #13600) Reported by: + atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 + (license 14) Tested by: atis + +2009-01-14 16:19 +0000 [r168598] Mark Michelson + + * channels/chan_agent.c: Fix a logic error I found while searching + through chan_agent.c I found that the allow_multiple_logins + function would never return 0 due to an incorrect comparison + being used when traversing the list of agents. While I was + modifying this function, I also did a little bit of coding + guidelines cleanup, too. + +2009-01-14 01:27 +0000 [r168593] Terry Wilson + + * apps/app_page.c: Don't overflow when paging more than 128 + extensions The number of available slots for calls in app_page + was hardcoded to 128. Proper bounds checking was not in place to + enforce this limit, so if more than 128 extensions were passed to + the Page() app, Asterisk would crash. This patch instead + dynamically allocates memory for the ast_dial structures and + removes the (non-functional) arbitrary limit. This issue would + have special importance to anyone who is dynamically creating the + argument passed to the Page application and allowing more than + 128 extensions to be added by an outside user via some external + interface. The patch posted by a_villacis was slightly modified + for some coding guidelines and other cleanups. Thanks, + a_villacis! (closes issue #14217) Reported by: a_villacis + Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch + uploaded by a (license 660) Tested by: otherwiseguy + +2009-01-13 19:13 +0000 [r168561] Russell Bryant + + * main/indications.c, main/channel.c, apps/app_read.c, + channels/chan_misdn.c, funcs/func_channel.c, + include/asterisk/indications.h, apps/app_disa.c, main/app.c, + res/snmp/agent.c, include/asterisk/channel.h, + res/res_indications.c: Revert unnecessary indications API change + from rev 122314 + +2009-01-13 18:34 +0000 [r168551] Terry Wilson + + * channels/chan_sip.c: Don't pass a value with a side effect to a + macro (closes issue #14176) Reported by: paraeco Patches: + chan_sip.c.diff uploaded by paraeco (license 658) + +2009-01-13 17:48 +0000 [r168546] Tilghman Lesher + + * funcs/func_logic.c: If either conditional is NULL, don't try + copying it. (closes issue #14226) Reported by: caspy Patches: + 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) + +2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler + + * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update + the app CDR field for AGI commands that are not executing an + application via "exec". + + * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG + Tested by: denisgalvao This gits rid of the notion of an + owning_app allowing the request and hangup to be initiated by + different threads. Originating from an active agent channel + requires this. The implementation primarily changes __login_exec + to wait on a condition variable rather than a lock. Review: + http://reviewboard.digium.com/r/35/ + +2009-01-12 14:58 +0000 [r168482] Mark Michelson + + * channels/chan_sip.c: I am reverting the fix made in revision + 168128 (and its upward merges) after being contacted by Olle + Johansson and being shown how this fix is incorrect. Thanks to + Olle for clearing this up for me. + +2009-01-12 14:57 +0000 [r168480] Russell Bryant + + * configs/indications.conf.sample: s/ringdance/ringcadence/ for + Bulgaria + +2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming + + * README: small commit to test new server + + * README: small commit to test new server + + * sounds/Makefile: update to use new sound file packages that + include license files + +2009-01-09 22:14 +0000 [r168198] Russell Bryant + + * res/res_musiconhold.c: Make this compile for mvanbaak + +2009-01-09 21:28 +0000 [r168191] Richard Mudgett + + * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 * + Miscellaneous doxygen comments added. + +2009-01-09 20:08 +0000 [r168128] Mark Michelson + + * channels/chan_sip.c: Add check_via calls to more request handlers + INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not + checking the topmost Via to determine where to send the response. + Adding check_via calls to those request handlers solves this. + (closes issue #13071) Reported by: baron Patches: check_via.patch + uploaded by baron (license 531) Tested by: baron + +2009-01-08 22:08 +0000 [r167840] Tilghman Lesher + + * res/res_agi.c: Don't truncate database results at 255 chars. + (closes issue #14069) Reported by: evandro Patches: + 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) + +2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming + + * channels/chan_sip.c: remove an unnecessary argument to + queue_request() + + * channels/chan_sip.c: When a SIP request or response arrives for a + dialog with an associated Asterisk channel, and the lock on that + channel cannot be obtained because it is held by another thread, + instead of dropping the request/response, queue it for later + processing when the channel lock becomes available. + http://reviewboard.digium.com/r/117/ + +2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant + + * main/file.c: Fix the last couple of places where free() was + improperly used directly. + + * main/file.c: Don't fclose() the file early, the filestream + destructor will handle it. + + * main/file.c: Only try to close the file if one was actually + opened + + * main/file.c: Don't use free() directly. This caused a crash since + ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, + #asterisk-dev + + * main/indications.c: Treat an empty string the same way as a NULL + country argument. In passing, simplify the handling of returning + a default tone zone. + +2009-01-06 21:35 +0000 [r167299] Mark Michelson + + * main/db.c: Use the correct variable when creating the format + string (closes issue #14177) Reported by: nic_bellamy Patches: + asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic + (license 299) + +2009-01-06 20:48 +0000 [r167260] Tilghman Lesher + + * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 + Jan 2009) | 2 lines Security fix AST-2009-001. ........ + +2009-01-05 16:51 +0000 [r167179] Mark Michelson + + * channels/chan_sip.c: A couple of changes to T.38 SDP attribute + handling There are some boolean attributes for T.38 such as + T38FaxFillBitRemoval, T38FaxTranscodingMMR, and + T38FaxTranscodingJBIG. By simply being present, we should treat + these as a "true" value. The current code, however, was requiring + a 1 or 0 as the value of the attribute in order to parse it. This + is due to the fact that there are some T.38 endpoints and + gateways that also transmit this information incorrectly. This + patch follows the "be liberal in what you accept and strict in + what you send" philosophy by accepting both the correctly- and + incorrectly-formatted attributes, but only sending information as + it is supposed to be sent. It was also discovered that a + particular type of T.38 gateway sends some non-standard T.38 SDP + attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, + it used T38MaxDatagram and T38FaxMaxRate respectively. We now + will properly accept these attributes as well. Note that there + are a lot of patches cited in the below commit message template. + This is because the person who submitted these patches is an + awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes + issue #13976) Reported by: linulin Patches: + chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov + (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded + by arcivanov (license 648) + chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov + (license 648) Tested by: arcivanov + +2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher + + * channels/chan_alsa.c: Repeat attempts to write when we receive + -EAGAIN from the driver, as detailed in the ALSA sample code (see + http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) + Reported by: Jerry Geis (via the -users list) Fixed by: me + (license 14) + + * channels/chan_local.c: Also inherit the musiconhold class. + (Closes #14153) Reported by: Jerry Geis, via the users list. + Patch by: me (license 14) + +2008-12-28 15:13 +0000 [r166772] Russell Bryant + + * channels/misdn_config.c: Use strncat() instead of an sprintf() in + which source and target buffers overlap + http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html + +2008-12-23 15:35 +0000 [r166592] Tilghman Lesher + + * main/asterisk.c, channels/chan_iax2.c: Compile, even if both + DAHDI and Zaptel are not installed. (Closes issue #14120) + +2008-12-23 15:16 +0000 [r166568] Mark Michelson + + * main/channel.c: Fix a crash resulting from a datastore with + inheritance but no duplicate callback The fix for this is to + simply set the newly created datastore's data pointer to NULL if + it is inherited but has no duplicate callback. (closes issue + #14113) Reported by: francesco_r Patches: 14113.patch uploaded by + putnopvut (license 60) Tested by: francesco_r + +2008-12-23 04:05 +0000 [r166509] Tilghman Lesher + + * main/channel.c: Use the integer form of condition for integer + comparisons. (closes issue #14127) Reported by: andrew + +2008-12-22 20:56 +0000 [r166380] Mark Michelson + + * channels/chan_dahdi.c: Fix a deadlock relating to channel locks + and autoservice It has been discovered that if a channel is + locked prior to a call to ast_autoservice_stop, then it is likely + that a deadlock will occur. The reason is that the call to + ast_autoservice_stop has a check built into it to be sure that + the thread running autoservice is not currently trying to + manipulate the channel we are about to pull out of autoservice. + The autoservice thread, however, cannot advance beyond where it + currently is, though, because it is trying to acquire the lock of + the channel for which autoservice is attempting to be stopped. + The gist of all this is that a channel MUST NOT be locked when + attempting to stop autoservice on the channel. In this particular + case, the channel was locked by a call to ast_read. A call to + ast_exists_extension led to autoservice being started and stopped + due to the existence of dialplan switches. It may be that there + are future commits which handle the same symptoms but in a + different location, but based on my looks through the code, it is + very rare to see a construct such as this one. (closes issue + #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded + by putnopvut (license 60) Tested by: rtrauntvein Review: + http://reviewboard.digium.com/r/107/ + +2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant + + * main/utils.c: Fix up timeout handling in ast_carefulwrite(). + + * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref + count handling of MoH classes using astobj2 to resolve crashes. + (closes issue #13566) Reported by: igorcarneiro Tested by: + russell Review: http://reviewboard.digium.com/r/106/ + +2008-12-19 23:34 +0000 [r166157] Mark Michelson + + * main/channel.c, funcs/func_audiohookinherit.c (added), + channels/chan_sip.c, include/asterisk/audiohook.h, + main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for + Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches: + 13538.patch uploaded by putnopvut (license 60) Tested by: + putnopvut + +2008-12-19 22:30 +0000 [r166093] Steve Murphy + + * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h, + apps/app_queue.c: This merges the masqpark branch into 1.4 These + changes eliminate the need for (and use of) the KEEPALIVE return + code in res_features.c; There are other places that use this + result code for similar purposes at a higher level, these appear + to be left alone in 1.4, but attacked in trunk. The reason these + changes are being made in 1.4, is that parking ends a channel's + life, in some situations, and the code in the bridge (and some + other places), was not checking the result code properly, and + dereferencing the channel pointer, which could lead to memory + corruption and crashes. Calling the masq_park function eliminates + this danger in higher levels. A series of previous commits have + replaced some parking calls with masq_park, but this patch puts + them ALL to rest, (except one, purposely left alone because a + masquerade is done anyway), and gets rid of the code that tests + the KEEPALIVE result, and the NOHANGUP_PEER result codes. While + bug 13820 inspired this work, this patch does not solve all the + problems mentioned there. I have tested this patch (again) to + make sure I have not introduced regressions. Crashes that + occurred when a parked party hung up while the parking party was + listening to the numbers of the parking stall being assigned, is + eliminated. These are the cases where parking code may be + activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to + parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip + xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via + manager. The interesting testing cases for parking are: I. A + calls B, A parks B a. B hangs up while A is getting the numbers + announced. b. B hangs up after A gets the announcement, but + before the parking time expires c. B waits, time expires, A is + redialed, A answers, B and A are connected, after which, B hangs + up. d. C picks up B while still in parking lot. II. A calls B, B + parks A a. A hangs up while B is getting the numbers announced. + b. A hangs up after B gets the announcement, but before the + parking time expires c. A waits, time expires, B is redialed, B + answers, A and B are connected, after which, A hangs up. d. C + picks up A while still in parking lot. Testing this throroughly + involves acting all the permutations of I and II, in situations + 1,2,3, and 4. Since I added a few more changes (ALL references to + KEEPALIVE in the bridge code eliimated (I missed one earlier), I + retested most of the above cases, and no crashes. H-extension + weirdness. Current h-extension execution is not completely + correct for several of the cases. For the case where A calls B, + and A parks B, the 'h' exten is run on A's channel as soon as the + park is accomplished. This is expected behavior. But when A calls + B, and B parks A, this will be current behavior: After B parks A, + B is hung up by the system, and the 'h' (hangup) exten gets run, + but the channel mentioned will be a derivative of A's... Thus, if + A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on + channel Parked/DAHDI/1-1, and the start/answer/end info + will be those relating to Channel A. And, in the case where A is + reconnected to B after the park time expires, when both parties + hang up after the joyful reunion, no h-exten will be run at all. + In the case where C picks up A from the parking lot, when either + A or C hang up, the h-exten will be run for the C channel. CDR's + are a separate issue, and not addressed here. As to WHY this + strange behavior occurs, the answer lies in the procedure + followed to accomplish handing over the channel to the parking + manager thread. This procedure is called masquerading. In the + process, a duplicate copy of the channel is created, and most of + the active data is given to the new copy. The original channel + gets its name changed to XXX and keeps the PBX + information for the sake of the original thread (preserving its + role as a call originator, if it had this role to begin with), + while the new channel is without this info and becomes a call + target (a "peer"). In this case, the parking lot manager thread + is handed the new (masqueraded) channel. It will not run an + h-exten on the channel if it hangs up while in the parking lot. + The h exten will be run on the original channel instead, in the + original thread, after the bridge completes. See bug 13820 for + our intentions as to how to clean up the h exten behavior. + Review: http://reviewboard.digium.com/r/29/ + +2008-12-19 19:48 +0000 [r165991] Jeff Peeler + + * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c, + apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c, + apps/app_dahdiscan.c, codecs/codec_dahdi.c, + res/res_musiconhold.c, channels/chan_iax2.c: (closes issue + #13480) Reported by: tzafrir Replace a bunch of if defined checks + for Zaptel/DAHDI through several new defines in dahdi_compat.h. + This removes a lot of code duplication. Example from bug: #ifdef + HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = + open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = + open(DAHDI_FILE_PSEUDO, O_RDRW); + +2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant + + * apps/app_chanspy.c: Ensure that the chanspy datastore is fully + initialized. This patch resolved some random crash issues + observed by a user on a BSD system (closes issue #14111) Reported + by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) + + * main/utils.c: Make ast_carefulwrite() be more careful. This patch + handles some additional cases that could result in partial writes + to the file description. This was done to address complaints + about partial writes on AMI. (issue #13546) (more changes needed + to address potential problems in 1.6) Reported by: srt Tested by: + russell Review: http://reviewboard.digium.com/r/99/ + +2008-12-18 21:14 +0000 [r165767] Tilghman Lesher + + * apps/app_voicemail.c: Add mutexes around accesses to the IMAP + library interface. This prevents certain crashes, especially when + shared mailboxes are used. (closes issue #13653) Reported by: + howardwilkinson Patches: + asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by + howardwilkinson (license 590) Tested by: jpeeler + +2008-12-18 18:52 +0000 [r165661] Russell Bryant + + * res/res_musiconhold.c: Set the process group ID on the MOH + process so that all children will get killed (closes issue + #14099) Reported by: caspy Patches: + res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license + 645) + +2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp + + * main/rtp.c: Only care about a compatible codec for early bridging + if we are actually bridging to another channel. If we are not we + actually want to bring the audio back to us. (closes issue + #13545) Reported by: davidw + + * apps/app_followme.c: Do not crash if we are not passed in a + followme id. (closes issue #14106) Reported by: ys Patches: + app_followme.c.2.diff uploaded by ys (license 281) + +2008-12-17 Russell Bryant + + * Asterisk 1.4.23-rc3 released. + +2008-12-17 21:14 +0000 [r165317] Tilghman Lesher + + * apps/app_macro.c: Reverse the fix from issue #6176 and add proper + handling for that issue. (Closes issue #13962, closes issue + #13363) Fixed by myself (license 14) + +2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson + + * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c, + apps/app_queue.c: Fix some memory leaks found while looking at + how realtime configs are handled. Also cleaned up some coding + guidelines violations in app_realtime.c, mostly related to + spacing + + * channels/chan_sip.c: After looking through SIP registration code + most of the day, this is one of the few things I could find that + was just plain wrong. Even though it probably isn't possible for + it to happen, it seems weird to have code that checks if a + pointer is NULL and then immediately dereferences that pointer if + it was NULL. + +2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant + + * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously + report that a thread is exiting while holding a lock. If the last + lock attempt was a trylock, and it failed, it will still be in + the list of locks so that it can be reported. (closes issue + #13219) Reported by: pj + + * apps/app_macro.c: Do not dereference the channel if + AST_PBX_KEEPALIVE has been returned. This is a bug I noticed + while looking at the code for app_macro. This return code means + that another thread has assumed ownership of the channel and it + can no longer be touched. (I hate this return code with a + passion, by the way.) + + * main/manager.c: Add "restart gracefully" to the AMI blacklist of + CLI commands. "module unload" was already identified as a command + that can not be used from the AMI. "restart gracefully" + effectively unloads all modules, and will run in to the same + problems. (closes issue #13894) Reported by: kernelsensei + + * include/asterisk/threadstorage.h, main/threadstorage.c: Fix + memory leak and invalid reporting issues with DEBUG_THREADLOCALS. + One issue was that the ast_mutex_* API was being used within the + context of the thread local data destructors. We would go off and + allocate more thread local data while the pthread lib was in the + middle of destroying it all. This led to a memory leak. Another + issue was an invalid argument being provided to the the + object_add API call. (closes issue #13678) Reported by: ys Tested + by: Russell + + * channels/chan_sip.c: Fix a memory leak related to the use of the + "setvar" configuration option. The problem was that these + variables were being appended to the list of vars on the sip_pvt + every time a re-registration or re-subscription came in. Since + it's just a waste of memory to put them there unless the request + was an INVITE, then the fix is to check the request type before + copying the vars. (closes issue #14037) Reported by: marvinek + Tested by: russell + +2008-12-16 15:15 +0000 [r164634] Steve Murphy + + * main/pbx.c: I added a sentence to clarify why - and ' ' are + ignored in patterns as per bug 14076. Leif says he'll put some + stuff about it in the extensions.conf sample, etc. + +2008-12-16 14:28 +0000 [r164605] Russell Bryant + + * res/res_musiconhold.c: Don't try to change working directory if a + directory was not configured. (closes issue #14089) Reported by: + caspy + +2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson + + * include/asterisk/pbx.h: Add the deadlock note to + ast_spawn_extension as well + + * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to + autoservice and pbx doxygen regarding a potential deadlock + scenario so that it is avoided in the future + +2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp + + * channels/chan_sip.c: Do not try to unlock a non-existant channel + if the transfer fails. (closes issue #13800) Reported by: dwagner + Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety + (license 608) + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/channel.h: Use autoconf logic to determine + whether the system has timersub or not. Do not blindly assume + Solaris does not. (closes issue #13838) Reported by: ano + + * apps/app_dial.c: Can we try not to assign an unsigned int to -1? + (closes issue #14074) Reported by: wetwired + +2008-12-15 14:31 +0000 [r164201] Russell Bryant + + * main/channel.c, res/res_features.c: Handle a case where a call + can be bridged to a channel that is still ringing. The issue that + was reported was about a case where a RINGING channel got + redirected to an extension to pick up a call from parking. Once + the parked call got taken out of parking, it heard silence until + the other side answered. Ideally, the caller that was parked + would get a ringing indication. This patch fixes this case so + that the caller receives ringback once it comes out of parking + until the other side answers. The fixes are: - Make sure we + remember that a channel was an outgoing channel when doing a + masquerade. This prevents an erroneous ast_answer() call on the + channel, which causes a bogus 200 OK to be sent in the case of + SIP. - Add some additional comments to explain related parts of + code. - Update the handling of the ast_channel visible_indication + field. Storing values that are not stateful is pointless. Control + frames that are events or commands should be ignored. - When a + bridge first starts, check to see if the peer channel needs to be + given ringing indication because the calling side is still + ringing. - Rework ast_indicate_data() a bit for the sake of + readability. (closes issue #13747) Reported by: davidw Tested by: + russell Review: http://reviewboard.digium.com/r/90/ + +2008-12-13 23:22 +0000 [r164082] Tilghman Lesher + + * apps/app_dial.c: Change the default calldurationlimit from the + special value 0 to -1, so we can better detect an exceptional + case. This follows on to the changes made in revision 156386. + Related to issue #13851. (closes issue #13974) Reported by: + paradise Patches: 20081208__bug13974.diff.txt uploaded by + Corydon76 (license 14) Tested by: file, blitzrage, ZX81 + +2008-12-12 22:20 +0000 [r163785] Russell Bryant + + * /: Set the reviewboard:url property on 1.4, as well + +2008-12-12 22:03 +0000 [r163761] Tilghman Lesher + + * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not + immediately exiting Asterisk, but also add a pointer inside + editline to look back to asterisk.c, so others don't spend as + much time as I did looking (in the wrong place) for the + appropriate function. Reported by: ZX81, via the #asterisk-users + channel Fixed by: me (license 14) + +2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant + + * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32 + so that it is actually 32 bits on 64-bit machines, as well. + (inspired by issue #13879) + + * main/channel.c, main/autoservice.c, include/asterisk/channel.h: + Resolve issues that could cause DTMF to be processed out of + order. These changes come from team/russell/issue_12658 1) Change + autoservice to put digits on the head of the channel's frame + readq instead of the tail. If there were frames on the readq that + autoservice had not yet read, the previous code would have + resulted in out of order processing. This required a new API call + to queue a frame to the head of the queue instead of the tail. 2) + Change up the processing of DTMF in ast_read(). Some of the + problems were the result of having two sources of pending DTMF + frames. There was the dtmfq and the more generic readq. Both were + used for pending DTMF in various scenarios. Simplifying things to + only use the frame readq avoids some of the problems. 3) Fix a + bug where a DTMF END frame could get passed through when it + shouldn't have. If code set END_DTMF_ONLY in the middle of digit + emulation, and a digit arrived before emulation was complete, + digits would get processed out of order. (closes issue #12658) + Reported by: dimas Tested by: russell, file Review: + http://reviewboard.digium.com/r/85/ + +2008-12-11 23:35 +0000 [r163383] Tilghman Lesher + + * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console, + on certain shells, the terminal is messed up. By intercepting + those events with a signal handler in the remote console, we can + avoid those issues. (closes issue #13464) Reported by: tzafrir + Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 + (license 14) Tested by: blitzrage + +2008-12-11 22:44 +0000 [r163316] Matt Nicholson + + * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes. + (closes issue #13819) Reported by: adomjan Patches: + pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) + dundi_clearecache3.diff uploaded by mnicholson (license 96) + Tested by: adomjan + +2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant + + * funcs/func_strings.c, funcs/func_cut.c: Fix some observed + slowdowns in dialplan processing. The change is to remove + autoservice usage from dialplan functions that do not need it + because they do not perform operations that potentially block. + (closes issue #13940) Reported by: tbelder + + * res/res_features.c: Fix an issue that made it so you could only + have a single caller executing a custom feature at a time. This + was especially problematic when custom features ran for any + appreciable amount of time. The fix turned out to be quite + simple. The dynamic features are now stored in a read/write list + instead of a list using a mutex. (closes issue #13478) Reported + by: neutrino88 Fix suggested by file + +2008-12-11 16:51 +0000 [r163088] Tilghman Lesher + + * res/res_agi.c: Don't wait forever, if there's a specified + recording timeout. (closes issue #13885) Reported by: bamby + Patches: res_agi.c.patch uploaded by bamby (license 430) + +2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson + + * apps/app_queue.c: Revert this cast to long. Using time_t here + causes build failures on a FreeBSD 32-bit build. + + * apps/app_queue.c: Fix a potential crash due to unsafe datastore + handling. This patch also contains a conversion from using long + to time_t for representing times for a queue, as well as some + whitespace fixes. (closes issue #14060) Reported by: nivek + Patches: datastore_fixup.patch.corrected uploaded by nivek + (license 636) with slight modification from me Tested by: nivek + +2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler + + * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp + check. Pointed out by mmichelson, thanks! + + * res/res_musiconhold.c: (closes issue #13229) Reported by: + clegall_proformatique Ensure that moh_generate does not return + prematurely before local_ast_moh_stop is called. Also, the sleep + in mp3_spawn now only occurs for http locations since it seems to + have been added originally only for failing media streams. + +2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp + + * channels/chan_sip.c: Fix subscription based MWI up a bit. We only + want to put sip: at the beginning of the URI if it is not already + there and revert code to ignore destination check if subscribing + for MWI. (closes issue #12560) Reported by: vsauer Patches: + patch001.diff uploaded by ramonpeek (license 266) + + * channels/chan_sip.c: When a SIP peer unregisters set the expiry + time back to 0 so that the 200 OK contains an expires of 0. + (closes issue #13599) Reported by: hjourdain Patches: + chan_sip.c.diff uploaded by hjourdain (license 583) + +2008-12-10 16:45 +0000 [r162671] Steve Murphy + + * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022) + Reported by: wetwired Tested by: murf I checked, and I added a + mod to the trunk version of Asterisk that would make it 8-bit + transparent on 27 Nov 2007, but I made no such updates to 1.4. My + best guess is that 1.4 was released, and it was not appropriate + to commit an enhancement. But I'm going to add the same fixes to + 1.4 now, for the following reasons: 1. wetwired is correct; 1.4 + is **mostly** 8-bit transparent now. This is because the lexical + token forming rules use . in most 'word' state continuances. It's + just the beginning of a 'word' that is picky. 2. Accepting 8-bit + chars in some places and not others leads to bug reports like + this. + +2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson + + * include/asterisk/stringfields.h: Update to stringfield handling + so that side-effects on parameters are not evaluated multiple + times. An example where this caused a problem was in chan_sip.c, + with the line ast_string_field_set(p, fromdomain, ++fromdomain); + This patch was originally uploaded to issue #13783 by jamessan. + While the issue was closed for other reasons, this patch is valid + and fixes a separate problem, and is thus being committed. + + * channels/chan_sip.c: Revert fix for issue 13570. It has caused + more problems than it helped to fix. (closes issue #13783) + Reported by: navkumar (closes issue #14025) Reported by: ffs + + * doc/misdn.txt: Add missing documentation to misdn.txt (closes + issue #14052) Reported by: festr Patches: misdn.txt.patch + uploaded by festr (license 443) + +2008-12-10 16:05 +0000 [r162653] Joshua Colp + + * main/rtp.c: Increment the sequence number on the end packets for + RFC2833. After reading the RFC some more and doing some testing I + agree with this change. (closes issue #12983) Reported by: vt + Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license + 520) + +2008-12-09 23:08 +0000 [r162463] Tilghman Lesher + + * apps/app_voicemail.c: Oops, should be "tz", not "zonetag". + +2008-12-09 22:17 +0000 [r162413] Russell Bryant + + * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove + the test_for_thread_safety() function completely. The test is not + valid. Besides, if we actually suspected that recursive mutexes + were not working, we would get a ton of LOG_ERROR messages when + DEBUG_THREADS is turned on. (inspired by a discussion on the + asterisk-dev list) + +2008-12-09 21:53 +0000 [r162348] Tilghman Lesher + + * apps/app_voicemail.c: We appear to have documented tz= in the + [general] section of voicemail.conf, without actually having + implemented it. Oops. (Reported by Olivier on the -users list) + +2008-12-09 21:14 +0000 [r162341] Joshua Colp + + * apps/app_directed_pickup.c: Add 'down' as a valid state for + directed call pickup. This creeps up when we receive session + progress when dialing a device and not ringing. (closes issue + #14005) Reported by: ddl + +2008-12-09 20:57 +0000 [r162286] Russell Bryant + + * apps/app_meetme.c: Fix an issue where callers on an incoming call + on an SLA trunk would not hear ringback. We need to make sure + that we don't start writing audio to the trunk channel until + we're actually ready to answer it. Otherwise, the channel driver + will treat it as inband progress, even though all they are + getting is silence. (closes issue #12471) Reported by: mthomasslo + +2008-12-09 20:44 +0000 [r162273] Joshua Colp + + * apps/app_festival.c: Fix double declaration of 'x' on the PPC + platform. (closes issue #14038) Reported by: ffloimair + +2008-12-09 20:28 +0000 [r162265] Mark Michelson + + * main/pbx.c: If we fail to start a thread for the pbx to run in, + we need to be sure to decrease the number of active calls on the + system. This fix may relate to ABE-1713, but it is not certain + yet. + +2008-12-09 20:20 +0000 [r162264] Steve Murphy + + * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with + seanbright on #asterisk-dev, I have added a default rule, and an + option to suppress the default rule from being generated in the + flex output, for the sake of those OS's where they didn't tweak + flex's ECHO macro, and the compiler doesn't like it. The + regressions are OK with this. + +2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp + + * main/rtp.c: Make sure that the timestamp for DTMF is not the same + as the previous voice frame and do not send audio when + transmitting DTMF as this confuses some equipment. (closes issue + #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file + (license 11) Tested by: ip-rob, bujones + + * main/rtp.c: Take video into account when early bridging RTP. + (closes issue #13535) Reported by: davidw + +2008-12-09 18:13 +0000 [r162136] Steve Murphy + + * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc + and ast_copy_string and messed up the standalone stuff. Fixed. + +2008-12-09 17:07 +0000 [r162071] Tilghman Lesher + + * channels/chan_phone.c: For some reason, after a distclean, gcc + started returning 'value computed is not used'. Fixing (for + --enable-dev-mode). + +2008-12-09 16:46 +0000 [r162014] Russell Bryant + + * apps/app_disa.c: Allow DISA to handle extensions that start with + #. (closes issue #13330) Reported by: jcovert + +2008-12-09 16:31 +0000 [r162013] Steve Murphy + + * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h, + pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme + Patches: 14019.diff uploaded by murf (license 17) Tested by: + ckjohnsonme, murf This crash was the result of a few small errors + that would combine in 64-bit land to result in a crash. 32-bit + land might have seen these combine to mysteriously drop the args + to an application call, in certain circumstances. Also, in trying + to find this bug, I spotted a situation in the flex input, where, + in passing back a 'word' to the parser, it would allocate a + buffer larger than necessary. I changed the usage in such + situations, so that strdup was not used, but rather, an + ast_malloc, followed by ast_copy_string. I removed a field from + the pval struct, in u2, that was never getting used, and set in + one spot in the code. I believe it was an artifact of a previous + fix to make switch cases work invisibly with extens. And, for + goto's I removed a '!' from before a strcmp, that has been there + since the initial merging of AEL2, that might prevent the proper + target of a goto from being found. This was pretty harmless on + its own, as it would just louse up a consistency check for users. + Many thanks to ckjohnsonme for providing a simplified and + complete set of information about the bug, that helped + considerably in finding and fixing the problem. Now, to get + aelparse up and running again in trunk, and out of its "horribly + broken" state, so I can run the regression suite! + +2008-12-09 14:52 +0000 [r161948] Russell Bryant + + * main/app.c: Fix a problem with GROUP() settings on a masquerade. + The previous code carried over group settings from the old + channel to the new one. However, it did nothing with the group + settings that were already on the new channel. This patch removes + all group settings that already existed on the new channel. I + have a more complicated version of this patch which addresses + only the most blatant problem with this, which is that a channel + can end up with multiple group settings in the same category. + However, I could not think of a use case for keeping any of the + group settings from the old channel, so I went this route for + now. (closes AST-152) + +2008-12-08 17:52 +0000 [r161725] Joshua Colp + + * channels/chan_sip.c: Make the usereqphone option work again. + (closes issue #13474) Reported by: mmaguire Patches: + 20080912_bug13474.diff uploaded by mmaguire (license 571) + +2008-12-05 21:02 +0000 [r161426] Sean Bright + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 161421 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec + 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned + int). (closes issue #14006) Reported by: alphaque Patches: + astobj2.h-patch uploaded by alphaque (license 259) (Slightly + modified by seanbright) ........ + +2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard + + * utils/smsq.c: kill a warning + +2008-12-05 14:12 +0000 [r161287] Russell Bryant + + * main/pbx.c: Fix a NULL format string warning found by buildbot. + +2008-12-04 18:30 +0000 [r161013] Jeff Peeler + + * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by: + jpeeler This mirrors a check that was present in ast_rtp_read to + also be in ast_rtp_raw_write to not schedule sending the receiver + report if the remote RTCP endpoint address isn't present in the + RTCP structure. Closes AST-142. + +2008-12-04 16:44 +0000 [r160943] Mark Michelson + + * main/callerid.c: Fix a callerid parsing issue. If someone + formatted callerid like the following: "name " (including + the quotation marks), then the parts would be parsed as name: + "name number: number This is because the closing quotation mark + was not discovered since the number and everything after was + parsed out of the string earlier. Now, there is a check to see if + the closing quote occurs after the number, so that we can know if + we should strip off the opening quote on the name. Closes AST-158 + +2008-12-03 21:54 +0000 [r160770] Tilghman Lesher + + * apps/app_voicemail.c: Some compilers warn on null format strings; + some don't (caught by buildbot) + +2008-12-03 21:38 +0000 [r160764] Jason Parker + + * channels/chan_agent.c: Only show this warning when we want to + show it. (closes issue #13982) Reported by: coolmig Patches: + chan_agent.c.patch uploaded by coolmig (license 621) + +2008-12-03 20:41 +0000 [r160703] Steve Murphy + + * funcs/func_callerid.c: (closes issue #13597) Reported by: + john8675309 Patches: patch.13597 uploaded by murf (license 17) + Tested by: murf, john8675309 This patch causes the setcid func to + update the CDR clid after setting the channel field. + +2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher + + * channels/chan_iax2.c: During bridge code, the channel bridge may + return a retry code, if a transfer was initiated but not yet + completed. If the bridge is immediately retried, then we may send + a storm of TXREQ packets, even though the first set is sent + reliably (retransmitted). Fixes AST-137. + + * pbx/pbx_spool.c: If an entry is added to the directory during a + scan when another entry expires, then that new entry will not be + processed promptly, but must wait for either a future entry to + start or a current entry's retry to occur. If no other entries + exist in the directory (other than the new entries) when a bunch + expire, then the new entries must wait until another new entry is + added to be processed. This was a rather weird race condition, + really. Fixes AST-147. + + * pbx/pbx_spool.c: Don't start scanning the directory until all + modules are loaded, because some required modules (channels, + apps, functions) may not yet be in memory yet. Fixes AST-149. + + * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev + list: "I guess that having only ip-phones in mind is not a good + approach. Since it is possible to have a sip proxy connected to + asterisk we could receive a 407 (unauthorized) or 483 (too many + hops) as response and dialog ending would not be a good + behavior." So modified. + +2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson + + * res/res_features.c: Channel is masqueraded, don't keep alive + + * res/res_features.c: A situation like A calls B, A builtin_atxfers + B to C, C parks B would lead to a crash. Thanks to file for + telling me how to fix it! (closes issue #13854) Reported by: Adam + Lee Tested by: otherwiseguy + +2008-12-02 17:42 +0000 [r160297] Tilghman Lesher + + * channels/chan_sip.c: When the text does not match exactly (e.g. + RTP/SAVP), then the %n conversion fails, and the resulting + integer is garbage. Thus, we must initialize the integer and + check it afterwards for success. (closes issue #14000) Reported + by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by + folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded + by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff + uploaded by folke (license 626) + +2008-12-02 01:16 +0000 [r160266] Terry Wilson + + * include/asterisk/astmm.h: make compile with dev mode and malloc + debug + +2008-12-02 00:25 +0000 [r160207] Tilghman Lesher + + * include/asterisk/stringfields.h, apps/app_voicemail.c, + main/pbx.c, main/frame.c: Ensure that Asterisk builds with + --enable-dev-mode, even on the latest gcc and glibc. + +2008-12-01 Tilghman Lesher + + * Released 1.4.23-rc2 + +2008-12-01 17:27 +0000 [r160003] Russell Bryant + + * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to + iax2_setoption(), as well, since they both have the potential to + send control frames in the middle of call setup. We have to wait + until we have received a message back from the remote end before + we try to send any more frames. Otherwise, the remote end will + consider it invalid, and we'll get stuck in an INVAL/VNAK storm. + +2008-12-01 16:08 +0000 [r159976] Michiel van Baak + + * main/manager.c: Get rid of the useless format string and argument + in the Bogus/ manager channelname. Noted by kpfleming and name + Bogus/manager suggested by eliel + +2008-12-01 14:52 +0000 [r159900] Russell Bryant + + * .cleancount: Force a "make clean" to avoid a bizarre build issue + ... + +2008-12-01 14:05 +0000 [r159897] Michiel van Baak + + * main/manager.c: make manager compile on OpenBSD. The last (10th) + argument to ast_channel_alloc here should be a pointer and NULL + is not really a pointer. + +2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming + + * main/enum.c, utils/frame.c, configure, res/res_agi.c, + include/asterisk/module.h, main/logger.c, main/dns.c, + include/asterisk/threadstorage.h, include/asterisk/utils.h, + include/asterisk/devicestate.h, channels/chan_sip.c, + include/asterisk/dundi.h, main/jitterbuf.c, + channels/chan_agent.c, configure.ac, utils/astman.c, + include/asterisk/cli.h, include/asterisk/channel.h, + include/jitterbuf.h, include/asterisk/manager.h, + main/ast_expr2.c, Makefile, include/asterisk/logger.h, + include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c, + include/asterisk/linkedlists.h, include/asterisk/lock.h, + include/asterisk/strings.h, makeopts.in, + include/asterisk/stringfields.h, utils/check_expr.c, + channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c: + update dev-mode compiler flags to match the ones used by default + on Ubuntu Intrepid, so all developers will see the same warnings + and errors since this branch already had some printf format + attributes, enable checking for them and tag functions that + didn't have them format attributes in a consistent way + +2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming + + * channels/chan_oss.c, channels/busy.h (removed), + channels/ring_tone.h (added), channels/chan_alsa.c, + channels/ringtone.h (removed), channels/busy_tone.h (added), + channels/Makefile: rename these files so as to avoid conflicts + when users update their working copies and have unversioned files + already in place + + * channels, agi/Makefile, utils/Makefile, channels/busy.h (added), + Makefile.moddir_rules, Makefile.rules, channels/ringtone.h + (added), channels/Makefile: simplify (and slightly bug-fix) the + recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h + and channels/ringtone.h to the repository instead of generating + them repeatedtly; most users do not change the settings to build + them, but the Makefile rules are still there if they wish to do + so ensure that 'make clean' removes dependency files for .i files + that are created in COMPILE_DOUBLE mode + +2008-11-25 22:41 +0000 [r159316] Steve Murphy + + * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported + by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) + Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX + fix. the change to cdr.c allows no-answer to percolate up into + CDR's, and feels like the right place to locate this fix; if BUSY + is done here, no-answer should be, too. + +2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher + + * channels/chan_iax2.c: Don't try to send a response on a NULL pvt. + (closes issue #13919) Reported by: barthpbx Patches: + chan_iax2.c.patch uploaded by eliel (license 64) Tested by: + barthpbx + + * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 + Nov 2008) | 7 lines Regression fix for last security fix. Set the + iseqno correctly. (closes issue #13918) Reported by: ffloimair + Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 + (license 14) Tested by: ffloimair ........ + +2008-11-25 17:34 +0000 [r159158] Russell Bryant + + * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to + go along with ao2_lock() and ao2_unlock() + +2008-11-25 16:23 +0000 [r159096] Terry Wilson + + * apps/app_festival.c: Add missing variable declaration in the PPC + code + +2008-11-25 04:50 +0000 [r159025] Tilghman Lesher + + * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: System call ioperm is non-portable, so check for + its existence in autoconf. (Closes issue #13863) + +2008-11-22 00:04 +0000 [r158629] Jeff Peeler + + * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes + issue #13786) Reported by: tzafrir When compiling against Zaptel + dahdi_compat will now only define all the DAHDI defines if the + Zaptel define is present. Also, there is no such thing as + DAHDI_PRI. + +2008-11-21 23:14 +0000 [r158603] Steve Murphy + + * res/res_features.c: In reference to the fix made for 13871, I was + merging the fix into 1.6.0 and realized I missed the code in the + h-exten block, and didn't catch it because my test case had the + h-exten commented out. So, this corrects the code I missed, as a + preventative against another crash report. Tested with the + h-exten defined, all is well. + +2008-11-21 23:07 +0000 [r158600] Tilghman Lesher + + * main/pbx.c: The passed extension may not be the same in the list + as the current entry, because we strip spaces when copying the + extension into the structure. Therefore, use the copied item to + place the item into the list. (found by lmadsen on -dev, fixed by + me) + +2008-11-21 22:05 +0000 [r158539] Russell Bryant + + * main/astobj2.c, include/asterisk/astobj2.h: When compiling with + DEBUG_THREADS, report the real file/func/line for + ao2_lock/ao2_unlock + +2008-11-21 21:19 +0000 [r158483] Steve Murphy + + * res/res_features.c: (closes issue #13871) Reported by: mdu113 + This one is totally my fault. The code doesn't even create a + bridge if the channel CDR has POST_DISABLED. I didn't check for + that at the end of the bridge. Fixed with a few small insertions. + Tested. Looks good. No cdr generated, no crash, no unnecc. data + objects created either. + +2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson + + * apps/app_queue.c: This change had somehow gotten reverted due to + a completely unrelated commit. Thanks to Theo Belder on the + Asterisk-dev list for pointing this out. + + * include/asterisk/file.h, main/frame.c, main/file.c, + include/asterisk/frame.h: There was an issue when attempting to + reference an embedded frame in a freed ast_filestream. This patch + makes use of the ao2 functions to make sure that we do not free + an ast_filestream structure until the embedded ast_frame has been + "freed" as well. (closes issue #13496) Reported by: fst-onge + Patches: filestream_frame_1_4.diff uploaded by putnopvut (license + 60) Tested by: putnopvut Closes AST-89 + + * channels/chan_sip.c: We don't handle 4XX responses to BYE well. + According to section 15 of RFC 3261, we should terminate a dialog + if we receive a 481 or 408 in response to our BYE. Since I am + aware of at least one phone manufacturer who may sometimes send a + 404 as well, I am being liberal and saying that any 4XX response + to a BYE should result in a terminated dialog. (closes issue + #12994) Reported by: pabelanger Patches: 12994.patch uploaded by + putnopvut (license 60) Closes AST-129 + + * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup + cause on the calling channel in the case that ast_call() fails. + For incoming SIP channels, this was causing us to send a 603 + instead of a 486 when the call-limit was reached on the + destination channel. (closes issue #13867) Reported by: still_nsk + Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: + blitzrage + +2008-11-20 01:46 +0000 [r158010] Richard Mudgett + + * channels/chan_misdn.c: Merged revision 157977 from + https://origsvn.digium.com/svn/asterisk/team/group/issue8824 + ........ Fixes JIRA ABE-1726 The dial extension could be empty if + you are using MISDN_KEYPAD to control ISDN provider features. + +2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming + + * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, + channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels, + main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, + codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, + channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules, + res/snmp, res/Makefile: the gcc optimizer frequently finds broken + code (use of uninitalized variables, unreachable code, etc.), + which is good. however, developers usually compile with the + optimizer turned off, because if they need to debug the resulting + code, optimized code makes that process very difficult. this + means that we get code changes committed that weren't adequately + checked over for these sorts of problems. with this build system + change, if (and only if) --enable-dev-mode was used and + DONT_OPTIMIZE is turned on, when a source file is compiled it + will actually be preprocessed (into a .i or .ii file), then + compiled once with optimization (with the result sent to + /dev/null) and again without optimization (but only if the first + compile succeeded, of course). while making these changes, i did + some cleanup work in Makefile.rules to move commonly-used + combinations of flag variables into their own variables, to make + the file easier to read and maintain + +2008-11-18 22:47 +0000 [r157503] Mark Michelson + + * channels/chan_sip.c: Add some missing invite state changes + necessary in the sip_write function. Not setting the invite state + correctly on the call was resulting in the Record application + leaving empty files. I also have updated the doxygen comment next + to the declaration of the INV_EARLY_MEDIA constant to reflect + that we also use this state when we *send* a 18X response to an + INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: + sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco + (license 162) Tested by: putnopvut + +2008-11-18 19:13 +0000 [r157365] Jeff Peeler + + * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel + This fix is the result of a bug fix in ast_app_separate_args + r124395. If an argument does not exist it should always be set to + a null string rather than a null pointer. + +2008-11-18 18:25 +0000 [r157305] Mark Michelson + + * apps/app_dial.c, channels/chan_local.c, res/res_features.c, + include/asterisk/channel.h, apps/app_followme.c: Fix a crash in + the end_bridge_callback of app_dial and app_followme which would + occur at the end of an attended transfer. The error occurred + because we initially stored a pointer to an ast_channel which + then was hung up due to a masquerade. This commit adds a "fixup" + callback to the bridge_config structure to allow for + end_bridge_callback_data to be changed in the case that a new + channel pointer is needed for the end_bridge_callback. + +2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming + + * Makefile, Makefile.rules: when an individual directory dist-clean + is run, run clean in that directory first, and when running + top-level dist-clean, do not run subdirectory clean operations + twice + + * Makefile.moddir_rules: dist-clean should remove dependency + information files as well + + * contrib/asterisk-ng-doxygen: major update to doxygen + configuration file: 1) update to doxygen 1.5.x style file, as + used in trunk 2) tell doxygen where are header files are, so + include-file processing can be done 3) make all macros that are + used to define variables/functions be expanded, so that doxygen + will properly document the resulting variable/function 4) make + all macros that are used to provide the contents of a variable + (structure) be expanded, so that doxygen will be able to document + the resulting fields 5) suppress compiler attributes + (__attribute__(xxx)) from being seen by doxygen, so it will + properly match up function definition and usage (for an example + of th effect of this, look at the doxygen docs for ast_log() from + before and afte this commit) + +2008-11-14 15:18 +0000 [r156816] Mark Michelson + + * apps/app_voicemail.c: If the prompt to reenter a voicemail + password timed out, it resulted in the password not being saved, + even if the input matched what you gave when first prompted to + enter a new password. This is because the return value of + ast_readstring was checked, but not checked properly. This bug + was discovered by Jared Smith during an Asterisk training course. + Thanks for reporting it! + +2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher + + * apps/app_while.c: ast_waitfordigit() requires that the channel be + up, for no good logical reason. This prevents While/EndWhile from + working within the "h" extension. Reported by: jgalarneau (for + ABE C.2) Fixed by: me + + * main/manager.c: Provide more space for all the data which can + appear in an originating channel name. (closes issue #13398) + Reported by: bamby Patches: manager.c.diff uploaded by bamby + (license 430) + +2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming + + * configure, autoconf/ast_gcc_attribute.m4: revert this change... + non-functional changes don't belong here + + * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax + error... no functional change + +2008-11-12 21:18 +0000 [r156386] Tilghman Lesher + + * apps/app_dial.c: When using call limits under 1 second, infinite + call lengths are allowed, instead. (closes issue #13851) Reported + by: ruddy + +2008-11-12 19:36 +0000 [r156297] Steve Murphy + + * main/pbx.c: It turns out that the 0x0XX00 codes being returned + for N, X, and Z are off by one, as per conversation with jsmith + on #asterisk-dev; he was teaching a class and disconcerted that + this published rule was not being followed, with patterns _NXX, + _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should + have been. This change, tested on these 3 patterns now picks the + proper one. However, this change may surprise users who set up + dialplans based on previous behavior, which has been there for + what, 2 and half years or so now. + +2008-11-12 19:26 +0000 [r156294] Tilghman Lesher + + * apps/app_meetme.c: If the SLA thread is not started, then reload + causes a memory leak. (closes issue #13889) Reported by: eliel + Patches: app_meetme.c.patch uploaded by eliel (license 64) + +2008-11-12 19:10 +0000 [r156289] Jeff Peeler + + * apps/app_meetme.c: For whatever reason, gcc only warned me about + the possible use of an uninitialized variable when compiling + 1.6.1. + +2008-11-12 18:39 +0000 [r156229] Tilghman Lesher + + * channels/chan_iax2.c: Revert revision 132506, since it + occasionally caused IAX2 HANGUP packets not to be sent, and + instead, schedule a task to destroy the iax2 pvt structure 10 + seconds later. This allows the IAX2 HANGUP packet to be queued, + transmitted, and ACKed before the pvt is destroyed. (closes issue + #13645) Reported by: dzajro Patches: + 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) + Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ + +2008-11-12 17:53 +0000 [r156178] Jeff Peeler + + * apps/app_meetme.c: (closes issue #13173) Reported by: pep This + change adds an announce_thread responsible for playing + announcements to an existing conference. This allows all + announcing to be immediately stopped if necessary but more + importantly allows other threads that need to play something to + not block. There are multiple benefits to this, but the actual + bug is for solving the scenario for a channel to be unusable + after hang up for the entire duration of the parting + announcement. The parting announcement can be extremely long + depending on what the user recorded upon joining the conference. + Reviewed by Russell on Review Board: + http://reviewboard.digium.com/r/25/ + +2008-11-12 17:38 +0000 [r156167] Mark Michelson + + * apps/app_dial.c: When doing some tests, I was having a crash at + the end of every call if an attended transfer occurred during the + call. I traced the cause to the CDR on one of the channels being + NULL. murf suggested a check in the end bridge callback to be + sure the CDR is non-NULL before proceeding, so that's what I'm + adding. + +2008-11-12 17:29 +0000 [r156164] Russell Bryant + + * main/asterisk.c: Move the sanity check that makes sure "always + fork" is not set along with the console option to be after the + code that reads options from asterisk.conf. This resolves a + situation where Asterisk can start taking up 100% when + misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting + me log in to his system to figure out what was causing the 100% + CPU problem.) + +2008-11-10 21:07 +0000 [r155861] Mark Michelson + + * channels/chan_agent.c: Channel drivers assume that when their + indicate callback is invoked, that the channel on which the + callback was called is locked. This patch corrects an instance in + chan_agent where a channel's indicate callback is called directly + without first locking the channel. This was leading to some + observed locking issues in chan_local, but considering that all + channel drivers operate under the same expectations, the generic + fix in chan_agent is the right way to go. AST-126 + +2008-11-10 20:49 +0000 [r155803] Tilghman Lesher + + * doc/valgrind.txt: I got tired of saying this in every single + bugnote referring to this file. + +2008-11-09 01:08 +0000 [r155553] Sean Bright + + * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h, + apps/app_followme.c: Use static functions here instead of nested + ones. This requires a small change to the ast_bridge_config + struct as well. To understand the reason for this change, see the + following post: + http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html + +2008-11-07 22:27 +0000 [r155398] Tilghman Lesher + + * channels/chan_sip.c: Clarify error message. (closes issue #13809) + Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded + by Corydon76 (license 14) Tested by: denke + +2008-11-06 19:45 +0000 [r155011] Mark Michelson + + * configs/voicemail.conf.sample: The documentation listed the + ability to set 'maxmsg' per context. The truth is that you can + only set this in the general section or per mailbox. Thus I am + updating the sample config file to be more accurate. Thanks to + sasargen on IRC for bringing up this issue. + +2008-11-05 16:44 +0000 [r154724] Mark Michelson + + * channels/chan_agent.c: The logic of a strcasecmp call was + reversed (closes issue #13841) Reported by: clegall_proformatique + +2008-11-05 16:06 +0000 [r154685] Steve Murphy + + * main/channel.c: This fix was prompted by communication from user, + who was seeing thousands of error logs... looks like EAGAIN. Made + such uninteresting. + +2008-11-04 20:49 +0000 [r154365] Tilghman Lesher + + * channels/chan_iax2.c: On busy systems, it's possible for the + values checked within a single line of code to change, unless the + structure is locked to ensure a consistent state. (closes issue + #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt + uploaded by Corydon76 (license 14) Tested by: kowalma + +2008-11-04 19:01 +0000 [r154266] Richard Mudgett + + * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to + the wrong state when it receives the indication + AST_CONTROL_RINGING. + +2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher + + * channels/chan_h323.c: Make the monitor thread non-detached, so it + can be joined (suggested by Russell on -dev list). + + * apps/app_voicemail.c: Attempting to expunge a mailbox when the + mailstream is NULL will crash Asterisk. (Closes issue #13829) + Reported by: jaroth Patch by: me (modified jaroth's patch) + + * main/rtp.c: Remove the potential for a division by zero error. + (Closes issue #13810) + +2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming + + * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c, + main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c, + res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in + this branch on the first pass + +2008-11-02 19:51 +0000 [r153651] Russell Bryant + + * include/asterisk/features.h: features.h depends on linkedlists.h, + so include it + +2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming + + * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c, + funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c, + channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c, + pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c, + pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c, + formats/format_wav.c, apps/app_festival.c, + main/db1-ast/hash/hash_page.c, main/translate.c, + res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c, + utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c, + pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h, + apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c, + utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c, + apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y, + apps/app_authenticate.c, formats/format_wav_gsm.c, + res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of + potential problems found by gcc 4.3.x, primarily bare strings + being passed to printf()-like functions and ignored results from + read()/write() and friends + +2008-10-31 22:36 +0000 [r153270] Terry Wilson + + * res/res_features.c, apps/app_followme.c: Add end_bridge_callback + for app_follome and add AUTOLOOP flag to res_features + +2008-10-31 Tilghman Lesher + + * Asterisk 1.4.23-rc1 released. + +2008-10-31 16:30 +0000 [r153114] Tilghman Lesher + + * channels/chan_sip.c: Turn off qualify on uncached realtime peers. + (Closes issue #13383) + +2008-10-31 15:45 +0000 [r153095] Terry Wilson + + * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h: + Recent CDR fixes moved execution of the 'h' exten into the + bridging code, so variables that were set after ast_bridge_call + was called would not show up in the 'h' exten. Added a callback + function to handle setting variables, etc. from w/in the bridging + code. Calls back into a nested function within the function + calling ast_bridge_call (closes issue #13793) Reported by: + greenfieldtech + +2008-10-30 20:58 +0000 [r152992] Sean Bright + + * bootstrap.sh: The -I argument to aclocal needs a space before the + include directory name. + +2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher + + * channels/chan_h323.c: Cannot join detached threads. See + http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html + (Closes issue #13400) + + * channels/chan_local.c: Unlock before returning, when extension + doesn't exist. (closes issue #13807) Reported by: eliel Patches: + chan_local.c.patch uploaded by eliel (license 64) + +2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming + + * main/cdr.c: instead of comparing the string pointer to 0, let's + compare the value that was actually parsed out of the string + (found by sparse) + +2008-10-29 05:23 +0000 [r152539] Russell Bryant + + * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes + issue #13795) Reported by: andrew53 Patches: + chan_sip_sizeof.patch uploaded by andrew53 (license 519) + +2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy + + * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c: + A little documentation cross-ref between features and dial and + queue... I wasted some time (stupidly) trying to get the + one-touch parking stuff working, because it didn't occur to me + that I had to also have the corresponding options in the dial + command! Duh! (In all this time, I never set this up before!) So, + to keep some poor fool from suffering the same fate, I made the + features.conf.sample file mention the corresponding opts in + dial/queue; and the docs for dial/app specifically mention the + corresponding decls in the feature.conf file. I hope this doesn't + spoil some vast, eternal plan... + + * apps/app_dial.c, res/res_features.c, funcs/func_channel.c, + include/asterisk/pbx.h, apps/app_queue.c: The magic trick to + avoid this crash is not to try to find the channel by name in the + list, which is slow and resource consuming, but rather to pay + attention to the result codes from the ast_bridge_call, to which + I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are + returned when a channel is parked. If you get AST_PBX_KEEPALIVE, + then don't touch the channel pointer. If you get + AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then + don't touch the peer pointer. Updated the several places where + the results from a bridge were not being properly obeyed, and + fixed some code I had introduced so that the results of the + bridge were not overridden (in trunk). All the places that + previously tested for AST_PBX_NO_HANGUP_PEER now have to check + for both AST_PBX_NO_HANGUP_PEER and + AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common + parking scenarios: 1. A calls B; B answers; A parks B; B hangs up + while A is getting the parking slot announcement, immediately + after being put on hold. 2. A calls B; B answers; A parks B; B + hangs up after A has been hung up, but before the park times out. + 3. A calls B; B answers; B parks A; A hangs up while B is getting + the parking slot announcement, immediately after being put on + hold. 4. A calls B; B answers; B parks A; A hangs up after B has + been hung up, but before the park times out. No crash. I also ran + the scenarios above against valgrind, and accesses looked good. + +2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher + + * apps/app_voicemail.c: Quoting in the wrong direction (Fixes + AST-107) + + * apps/app_dial.c: Reset all DIAL variables back to blank, in case + Dial is called multiple times per call (which could otherwise + lead to inconsistent status reports). (closes issue #13216) + Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded + by Corydon76 (license 14) Tested by: ruddy + +2008-10-27 23:28 +0000 [r152286] Jeff Peeler + + * channels/chan_dahdi.c: Buffer policy setting for half is not + needed. + +2008-10-27 21:32 +0000 [r152215] Tilghman Lesher + + * channels/chan_local.c: Inherit ALL elements of CallerID across a + local channel. (closes issue #13368) Reported by: Peter Schlaile + Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 + (license 14) + +2008-10-26 20:23 +0000 [r152059] Sean Bright + + * funcs/func_strings.c: Since passing \0 as the second argument to + strchr is valid (and will match the trailing \0 of a string) we + need to check that first, otherwise we end up with incorrect + results. Fix suggested by reporter. (closes issue #13787) + Reported by: meitinger + +2008-10-25 10:59 +0000 [r151905] Russell Bryant + + * main/asterisk.c: Move AMI initialization to occur after loading + modules. This prevents a deadlock when someone tries to initiate + a module reload from the AMI just as Asterisk is starting. + (closes issue #13778) Reported by: hotsblanc Fix suggested by + hotsblanc + +2008-10-23 16:04 +0000 [r151763] Terry Wilson + + * configs/features.conf.sample, res/res_features.c, CHANGES: + Backport fix from 1.6.0 that allows you to set + parkedcalltransfers=no|caller|callee|both, but default to both + which would be the equivalent of the existing behaviour. The + problem was that if someone parked a call, the callee and caller + would both get assigned the builtin transfer feature, which would + not only be potentially giving someone the ability to transfer + themselves when they shouldn't have it, but would also dissallow + reinviting the media off of the call. (closes issue #12854) + Reported by: davidw Patches: parkingfix4.diff.txt uploaded by + otherwiseguy Tested by: davidw, otherwiseguy + +2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming + + * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4, + configure.ac: rename this macro to properly reflect what it does + + * autoconf/ast_check_pwlib.m4 (added), autoconf (added), + autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4 + (added), configure, autoconf/ast_gcc_attribute.m4 (added), + bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added), + autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added), + autoconf/ast_c_compile_check.m4 (added), + autoconf/ast_c_define_check.m4 (added), + autoconf/ast_prog_egrep.m4 (added), + autoconf/ast_check_openh323.m4 (added), + autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4 + (added), acinclude.m4 (removed): break up acinclude.m4 into + individual files, which will make it easier to maintain, easier + to add new macros (less patching) and will ease maintenance of + these macros across Asterisk branches + +2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke + + * main/asterisk.c: As per kpfleming's comments to the prior commit, + I'm reverting some of the changes here. A comment was made in bug + #13726 "3. The same mistake as in (2) is done in a few other + places in the code that check for: #if defined(HAVE_ZAPTEL) || + defined(HAVE_DAHDI) Harmless, but still incorrect." In the case + of main/asterisk.c, this is not incorrect because without + HAVE_ZAPTEL defined, we're missing the include for ioctl and the + namespace that defines DAHDI_TIMERCONFIG which is still required + when using Zaptel with the 1.4 branch. + + * main/asterisk.c: Fix the 1.4 branch compile again broken with + r150557 when using with Zaptel and not DAHDI (closes issue + #13740) reported by: jmls patch by: bweschke + +2008-10-18 01:42 +0000 [r150816] BJ Weschke + + * main/manager.c: Using the GetVar handler in AMI is potentially + dangerous (insta-crash [tm]) when you use a dialplan function + that requires a channel and then you don't provide one or provide + an invalid one in the Channel: parameter. We'll handle this + situation exactly the same way it was handled in pbx.c back on + r61766. We'll create a bogus channel for the function call and + destroy it when we're done. If we have trouble allocating the + bogus channel then we're not going to try executing the function + call at all and run the risk of crashing. (closes issue #13715) + reported by: makoto patch by: bweschke + +2008-10-17 17:18 +0000 [r150637] Steve Murphy + + * res/res_features.c: Interesting crash. In this case, you exit the + bridge with peer completely GONE. I moved the channel find call + up to cover the whole peer CDR reset code segment. This appears + to solve the crash without changing the logic at all. + +2008-10-17 15:31 +0000 [r150557] Jason Parker + + * main/asterisk.c, main/channel.c, channels/chan_dahdi.c, + configure, configure.ac: Correctly allow chan_dahdi to compile + against older versions of Zaptel. Don't always define + HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor + cleanup to make things clear. (closes issue #13726) Reported by: + tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46) + +2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson + + * main/manager.c: Reverting changes from commits 150298 and 150301 + since I was mistakenly under the assumption that dialplan + functions *always* required that a channel be present. I need to + go home earlier, I think :) + + * main/manager.c: And don't forget to return on the error condition + + * main/manager.c: Don't try to call a dialplan function's read + callback from the manager's GetVar handler if an invalid channel + has been specified. Several dialplan functions, including CHANNEL + and SIP_HEADER, do not check for NULL-ness of the channel being + passed in. (closes issue #13715) Reported by: makoto + +2008-10-16 15:56 +0000 [r150124] Richard Mudgett + + * channels/chan_misdn.c: Fix memory leak found by customer + +2008-10-16 15:26 +0000 [r150056] Steve Murphy + + * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and + RYM-150398 and AST-103 in internal Digium bug trackers. These + fixes address a really subtle memory corruption problem that + would happen in machines heavily loaded in production + environments. The corruption would always take the form of the + STMT object getting nulled out and one of the unixODBC calls + would crash trying to access statement->connection. It isn't + fully proven yet, but the server has now been running 2.5 days + without appreciable memory growth, or any gain of %cpu, and no + crashes. Whether this is the problem or not on that server, these + fixes are still warranted. As it turns out, **I** introduced + these errors unwittingly, when I corrected another crash earlier. + I had formed the build_query routine, and failed to remove + mutex_unlock calls in 3 places in the transplanted code. These + unlocks would only happen in error situations, but unlocking the + mutex early set the code up for a catastrophic failure, it + appears. It would happen only once every 100K-200K or more calls, + under heavy load... but that is enough. If another crash occurs, + with the same MO, I'll come back and remove my confession from + the log, and we'll keep searching, but the fact that we have + Asterisk dying from an asynchronous wiping of the STMT object, + only on some connection error, and that the server has lived for + 2.5 days on this code without a crash, sure make it look like + this was the problem! Also, in several points, Statement handles + are set to NULL after SQLFreeHandle. This was mainly for + insurance, to guarantee a crash. As it turns out, the code does + not appear to be attempting to use these freed pointers. Asterisk + owes a debt of gratitude to Federico Alves and Frediano Ziglio + for their untiring efforts in finding this bug, among others. + +2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke + + * CHANGES: Another documentation fix. (closes issue #13708) + + * configs/agents.conf.sample: An update to the + documentation/example of agents.conf.sample with the correct + parameter for this feature as defined in chan_agent.c (closes + issue #13709) + +2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming + + * channels/chan_sip.c: fix some problems when parsing SIP messages + that have the maximum number of headers or body lines that we + support + +2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson + + * channels/chan_sip.c: Change this warning to an error message. + Suggestion comes from Sean Bright. Thanks Sean! + + * channels/chan_sip.c: Call register_peer_exten even in the case + that the peer's IP/port does not change. (closes issue #13309) + Reported by: dimas Patches: v2-13309.patch uploaded by dimas + (license 88) + + * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance + period for sync-triggered audiohooks so that if packetization of + audio is close (but not equal) we don't end up flushing the + audiohooks over small inconsistencies in synchronization. Related + to issue #13005, and solves the issue for most people who were + experiencing the problem. However, a small number of people are + still experiencing the problem on long calls, so I am not closing + the issue yet + + * apps/app_queue.c: Update the queue with the correct number of + calls and whether the call was completed within the service level + when a transfer takes place. This way, we do not "break" the + leastrecent and fewestcalls strategies by not logging a call + until after the transferred call has ended. (closes issue #13395) + Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded + by Marquis (license 32) + + * channels/chan_sip.c: Don't allow reserved characters to be used + in register lines in sip.conf. (closes issue #13570) Reported by: + putnopvut + +2008-10-14 20:09 +0000 [r149061] Tilghman Lesher + + * apps/app_waitforsilence.c: Check correct values in the return of + ast_waitfor(); also, get rid of a possible memory leak. (closes + issue #13658) Reported by: explidous Patch by: me + +2008-10-14 19:05 +0000 [r148990] Leif Madsen + + * CHANGES: Add in some missing updates to the CHANGES file for + sip.conf (closes issue #13100) Reported and patch by: + gknispel_proformatique + +2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher + + * apps/app_voicemail.c: Some compilers warn, some don't. Fixing. + + * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean, + even when UTF-8 characters are used in headers like 'Subject' and + 'To'. Closes AST-107. + +2008-10-14 17:33 +0000 [r148912] Mark Michelson + + * channels/chan_local.c: Deadlock prevention in chan_local. (closes + issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded + by putnopvut (license 60) Tested by: tacvbo + +2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming + + * Makefile: on Ubuntu (at least), recent versions of ld in binutils + delete all debugging symbols when -x is supplied; since the + reasons why -x is being passed are lost in the mists of time, + remove it so debugging will work properly + + * main/translate.c: it would be nice if this message printing code + had actually been tested before it was committed... + +2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher + + * apps/app_voicemail.c: User not notified of temporary greeting, if + ODBC storage is in use. (closes issue #13659) Reported by: + moliveras Patches: 20081009__bug13659.diff.txt uploaded by + Corydon76 (license 14) Tested by: moliveras + + * apps/app_voicemail.c: When blank, callerid name and number should + display "unknown caller" in voicemail emails. (Closes issue + #13643) + +2008-10-09 18:56 +0000 [r147941] Jeff Peeler + + * res/res_features.c: (closes issue #13139) Reported by: krisk84 + Tested by: krisk84 This change prevents a call that is placed in + the parkinglot to be picked up before the PBX is finished. If + another extension dials the parking extension before the PBX + thread has completed at minimum warnings will occur about the PBX + not properly being terminated. At worst, a crash could occur. + +2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming + + * channels/chan_dahdi.c: when parsing a text configuration option, + ensure that the buffer on the stack is actually large enough to + hold the legal values of that option, and also ensure that + sscanf() knows to stop parsing if it would overrun the buffer + (without these changes, specifying "buffers=...,immediate" would + overflow the buffer on the stack, and could not have worked as + expected) + +2008-10-08 14:51 +0000 [r147517] Joshua Colp + + * apps/app_speech_utils.c: If we receive DTMF make sure that the + state of the speech structure goes back to being not ready. + (issue #LUMENVOX-8) + +2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming + + * channels/chan_dahdi.c: revert this change until i can understand + why it results in locking order changes + + * channels/chan_dahdi.c: don't start a PBX on incoming PRI call + channels until after we're done setting channel variables and + other things on the channel, otherwise the channel might go away + (if the dialplan hangs up quickly) before we are done, which + results in a spectacular crash + +2008-10-07 16:48 +0000 [r147193] Sean Bright + + * apps/app_voicemail.c: Make 'imapsecret' an alias to + 'imappassword' in voicemail.conf. + +2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher + + * funcs/func_callerid.c, apps/app_speech_utils.c, + funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, + channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c, + funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c: + Dialplan functions should not actually return 0, unless they have + modified the workspace. To signal an error (and no change to the + workspace), -1 should be returned instead. (closes issue #13340) + Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt + uploaded by Corydon76 (license 14) + + * channels/chan_local.c: Check whether an extension exists in the + _call method, rather than the _alloc method, because we need to + evaluate the callerid (since that data affects whether an + extension exists). (closes issue #13343) Reported by: efutch + Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 + (license 14) Tested by: efutch + +2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming + + * channels/chan_dahdi.c: ensure that the private structure for + pseudo channels is created without 'leaking' configuration data + from other configured channels (closes issue #13555) Reported by: + jeffg Patches: issue_13555.patch uploaded by kpfleming (license + 421) Tested by: jeffg + +2008-10-05 21:17 +0000 [r146448] Jason Parker + + * channels/chan_sip.c: Fix silly formatting. + +2008-10-03 22:51 +0000 [r146244] Sean Bright + + * apps/app_rpt.c: Change some preprocessor macros to struct + definitions so that we get app_rpt to build with DAHDI. (closes + issue #13576) Reported by: blitzrage + +2008-10-03 20:44 +0000 [r146129] Jeff Peeler + + * include/asterisk/features.h, res/res_features.c, res/res_agi.c: + (closes issue #13425) Reported by: mdu113 Tested by: mdu113 + Similar to r143204, masquerade the channel in the case of Park + being called from AGI. + +2008-10-03 17:12 +0000 [r146026] Steve Murphy + + * res/res_features.c: (closes issue #13579) Reported by: dwagner + (closes issue #13584) Reported by: dwagner Tested by: murf, + putnopvut The thought occurred to me that the res= from the + extension spawn was ending up being returned from the bridge. + "Thou shalt not poison the return value". Made the change and it + appears to allow blind xfers to work as normal. If I'm wrong, + reopen the bugs. But it looks good to me! Many thanks to + putnopvut for helping me reproduce this! + +2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher + + * funcs/func_odbc.c: Backport support for some of the keyword + modifications used in 1.6 (while warning that some options aren't + really supported) and add some warning messages. Some credit to + oej, who was complaining in #asterisk-dev. + + * res/res_odbc.c: Some sanity checks that may have led to prior + crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup + of incorrectly-used constants. + +2008-10-01 17:18 +0000 [r145479] Leif Madsen + + * contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql + script to create the setinterfacevar column. (closes issue + #13549) Reported by: fiddur + +2008-10-01 Russell Bryant + + * Asterisk 1.4.22 released. + +2008-09-09 Russell Bryant + + * Asterisk 1.4.22-rc5 released. + +2008-09-09 15:40 +0000 [r142063] Russell Bryant + + * res/res_features.c: Ensure that the stored CDR reference is still + valid after the bridge before poking at it. Also, keep the + channel locked while messing with this CDR. (fixes crashes + reported in issue #13409) + +2008-09-08 21:10 +0000 [r141809] Mark Michelson + + * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check + the remote tag of an endpoint once a dialog has been confirmed. + Up until that point, it is possible and legal for the far-end to + send provisional responses with a different To: tag each time. + With this patch applied, these provisional messages will not + cause a matching problem. (closes issue #11536) Reported by: ibc + Patches: 11536v2.patch uploaded by putnopvut (license 60) + +2008-09-08 21:02 +0000 [r141806] Russell Bryant + + * main/pbx.c: When doing an async goto, detect if the channel is + already in the middle of a masquerade. This can happen when + chan_local is trying to optimize itself out. If this happens, + fail the async goto instead of bursting into flames. (closes + issue #13435) Reported by: geoff2010 + +2008-09-08 Russell Bryant + + * Asterisk 1.4.22-rc4 released. + +2008-09-08 20:15 +0000 [r141741] Jason Parker + + * Makefile, redhat (removed): Remove RPM package targets from + Makefile (and all associated parts). This has never worked in + 1.4, and we decided that it makes no sense to be done here. There + are many distros out there that already have "proper" spec files + that can be (re)used. Closes issue #13113 Closes issue #10950 + Closes issue #10952 + +2008-09-08 16:26 +0000 [r141678] Russell Bryant + + * configure, configure.ac: Actually use Zaptel CFLAGS if using + Zaptel instead of DAHDI This fixes building against Zaptel when + using a custom path + +2008-09-06 20:13 +0000 [r141565] Steve Murphy + + * channels/chan_sip.c: This fix comes from Joshua Colp The + Brilliant, who, given the trace, came up with a solution. This + will most likely will close 13235 and 13409. I'll wait till + Monday to verify, and then close these bugs. + +2008-09-06 15:23 +0000 [r141503] Tilghman Lesher + + * res/res_agi.c: Reverting behavior change (AGI should not exit + non-zero on SUCCESS) (closes issue #13434) Reported by: + francesco_r + +2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson + + * channels/chan_agent.c: Agent's should not try to call a channel's + indicate callback if the channel has been hung up. It will likely + crash otherwise ABE-1159 + + * apps/app_voicemail.c: Since greetings are not stored in IMAP, we + should not be DISPOSE'ing of them the same way we do with other + messages. (closes issue #13414) Reported by: mthomasslo Patches: + 13414v2.patch uploaded by putnopvut (license 60) Tested by: + mthomasslo + + * channels/chan_sip.c: Commit 140417 had a logic flaw in it which + caused port 5060 to always be used when dialing a peer if no + explicit port was specified. This broke the behavior of + implicitly using the port from which the peer registered if no + port is specified. This commit fixes the logic flaw. (closes + issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded + by putnopvut (license 60) Tested by: mdu113 + +2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy + + * main/channel.c: A small change to prevent double-posting of + CDR's; thanks to Daniel Ferrer for bringing it to our attention + + * pbx/ael/ael-test/ref.ael-vtest25 (added), + pbx/ael/ael-test/ael-vtest25/extensions.ael (added), + pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c, + pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue + #13357) Reported by: pj Tested by: murf (closes issue #13416) + Reported by: yarns Tested by: murf If you find this message + overly verbose, relax, it's probably not meant for you. This + message is meant for probably only two people in the whole world: + me, or the poor schnook that has to maintain this code because + I'm either dead or unavailable at the moment. This fix solves two + reports, both having to do with embedding a function call in a + ${} construct. It was tricky because the funccall syntax has + parenthesis () in it. And up till now, the 'word' token in the + flex stuff didn't allow that, because it would tend to steal the + LP and RP tokens. To be truthful, the "word" token was the + trickiest, most unstable thing in the whole lexer. I was lucky it + made this long without complaints. I had to choose every + character in the pattern with extreme care, and I knew that + someday I'd have to revisit it. Well, the day has come. So, my + brilliant idea (and I'm being modest), was to use the surrounding + ${} construct to make a state machine and capture everything in + it, no matter what it contains. But, I have to now treat the word + token like I did with comments, in that I turn the whole thing + into a state-machine sort of spec, with new contexts + "curlystate", "wordstate", and "brackstate". Wait a minute, + "brackstate"? Yes, well, it didn't take very many regression + tests to point out if I do this for ${} constructs, I also have + to do it with the $[] constructs, too. I had to create a separate + pcbstack2 and pcbstack3 because these constructs can occur inside + macro argument lists, and when we have two state machines + operating on the same structures we'd get problems otherwise. I + guess I could have stopped at pcbstack2 and had the brackstate + stuff share it, but it doesn't hurt to be safe. So, the pcbpush + and pcbpop routines also now have versions for "2" and "3". I had + to add the {KEYWORD} construct to the initial pattern for "word", + because previously word would match stuff like "default7", + because it was a longer match than the keyword "default". But, + not any more, because the word pattern only matches only one or + two characters now, and it will always lose. So, I made it the + winner again by making an optional match on any of the keywords + before it's normal pattern. I added another regression test to + make sure we don't lose this in future edits, and had to fix just + one regression, where it no longer reports a 'cascaded' error, + which I guess is a plus. I've given some thought as to whether to + apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I + decided to put it in 1.4 because one of the bug reports was + against 1.4; and it is unexpected that AEL cannot handle this + situation. It actually reduced the amount of useless "cascade" + error messages that appeared in the regressions (by one line, + ehhem). There is a possible side-effect in that it does now do + more careful checking of what's in those ${} constructs, as far + as matching parens, and brackets are concerned. Some users may + find a an insidious problem and correct it this way. This should + be exceedingly rare, I hope. + +2008-09-04 17:00 +0000 [r141028] Jeff Peeler + + * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes + multiple parking problems: Crash when executing a park on an + extension dialed by AGI due to not returning the proper return + code. Crash when using a builtin feature that was a subset of a + enabled dynamic feature. Crash due to always hanging up the peer + despite the fact that the peer was supposed to be parked. + +2008-09-03 Russell Bryant + + * Asterisk 1.4.22-rc3 released. + +2008-09-03 14:29 +0000 [r140850] Mark Michelson + + * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC + storage. (closes issue #13387) Reported by: moliveras Patches: + 13387.patch uploaded by putnopvut (license 60) Tested by: + putnopvut, moliveras + +2008-09-03 13:24 +0000 [r140816] Russell Bryant + + * main/poll.c: Don't freak out if the poll emulation receives NULL + for the pollfds array (closes issue #13307) Reported by: jcovert + +2008-09-02 23:47 +0000 [r140751] Mark Michelson + + * apps/app_voicemail.c: After adding the context checking to + app_voicemail for IMAP storage, I left out a crucial place to + copy the context to the vm_state structure. This is the + correction. + +2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy + + * main/cdr.c: I am turning the warnings generated in ast_cdr_free + and post_cdr into verbose level 2 messages. Really, they matter + little to end users. You either get the CDR's you wanted, or you + don't, and it is a bug. + + * main/channel.c: After reconsidering, with respect to 13409, + ast_cdr_detach should be OK, better in fact, than ast_cdr_free, + which generates lots of useless warnings that will undoubtably + generate complaints. + + * main/channel.c, main/pbx.c: (closes issue #13409) Reported by: + tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by + tomaso (license 564) I basically spent the day, verifying that + this patch solves the problem, and doesn't hurt in non-problem + cases. Why valgrind did not plainly reveal this leak absolutely + mystifies and stuns me. Many, many thanks to tomaso for finding + and providing the fix. + +2008-09-02 18:14 +0000 [r140605] Sean Bright + + * channels/chan_iax2.c: Make sure to use the correct length of the + mohinterpret and mohsuggest buffers when copying configuration + values. (closes issue #13336) Reported by: + decryptus_proformatique Patches: + chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded + by decryptus (license 555) + +2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson + + * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After + working on the ao2_containers branch, I noticed something a bit + strange. In all cases where we provide a callback function to + ao2_container_alloc, the callback function would only return 0 or + CMP_MATCH. After inspecting the ao2_callback() code carefully, I + found that if you're only looking for one specific item, then you + should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will + continue traversing the current bucket until the end searching + for more matches. In cases like chan_iax2 where in 1.4, all the + peers are shoved into a single bucket, this makes for potentially + terrible performance since the entire bucket will be traversed + even if the peer is one of the first ones come across in the + bucket. All the changes I have made were for cases where the + callback function defined was passed to ao2_container_alloc so + that calls to ao2_find could find a unique instance of whatever + object was being stored in the container. + + * apps/app_voicemail.c: Add context checking when retrieving a + vm_state. This was causing a problem for people who had + identically named mailboxes in separate voicemail contexts. This + commit affects IMAP storage only. (closes issue #13194) Reported + by: moliveras Patches: 13194.patch uploaded by putnopvut (license + 60) Tested by: putnopvut, moliveras + + * channels/chan_sip.c: Fix SIP's parsing so that if a port is + specified in a string to Dial(), it is not ignored. (closes issue + #13355) Reported by: acunningham Patches: 13355v2.patch uploaded + by putnopvut (license 60) Tested by: acunningham + +2008-08-27 19:49 +0000 [r140299] Mark Michelson + + * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked + when in pedantic mode. The problem was that the wrong tags would + be compared depending on the direction of the call. (closes issue + #13353) Reported by: flefoll Patches: + chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll + (license 244) + +2008-08-26 16:49 +0000 [r140115] Jeff Peeler + + * channels/chan_dahdi.c: add HAVE_PRI if define around + dahdi_close_pri_fd + +2008-08-26 16:07 +0000 [r140060] Russell Bryant + + * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip. + This code used the return value of a completely unrelated + function to determine whether the scheduler should be run or not. + This would have caused the scheduler to not run in cases where it + should have. Also, leave a note about another scheduler issue + that needs to be addressed at some point. + +2008-08-26 15:57 +0000 [r140056] Jeff Peeler + + * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir + Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested + by: tzafrir, jpeeler This patch fixes closing open file + descriptors in the case of an error. + +2008-08-26 15:27 +0000 [r140051] Russell Bryant + + * channels/chan_iax2.c: Fix a race condition with the IAX scheduler + thread. A lock and condition are used here to allow newly + scheduled tasks to wake up the scheduler just in case the new + task needs to run sooner than the current wakeup time when the + thread is sleeping. However, there was a race condition such that + a newly scheduled task would not properly wake up the scheduler + or affect the wake up period. The order of execution would have + been: 1) Scheduler thread determines wake up time of N ms. 2) + Another thread schedules a task and signals the condition, with + an execution time of < N ms. 3) Scheduler thread locks and goes + to sleep for N ms. By moving the sleep time determination to + inside the critical section, this possibility is avoided. + +2008-08-26 15:22 +0000 [r140050] Terry Wilson + + * Makefile: sounds/Makefile installs sounds using the "new" + language directory structure, but languageprefix needs to be set + = yes for sounds in subdirectories (digits/1, etc.) to play as + the correct language. Fix the generation of asterisk.conf to + include languageprefix=yes + +2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming + + * channels/chan_dahdi.c: correct a file location in an error + message + +2008-08-25 21:47 +0000 [r139927] Jeff Peeler + + * main/manager.c: Fix a typo I made. Lesson learned, apply the + patch if one exists. + +2008-08-25 21:31 +0000 [r139909] Sean Bright + + * build_tools/get_moduleinfo, build_tools/get_makeopts: Some + versions of awk (nawk, for example) don't like empty regular + expressions so be slightly more verbose. (closes issue #13374) + Reported by: dougm Patches: 13374.diff uploaded by seanbright + (license 71) Tested by: dougm + +2008-08-25 20:46 +0000 [r139869] Terry Wilson + + * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely + +2008-08-25 15:52 +0000 [r139769] Mark Michelson + + * main/config.c: Fix the logic in config_text_file_save so that if + an UpdateConfig manager action is issued and the file specified + in DstFileName does not yet exist, an error is not returned. + (closes issue #13341) Reported by: vadim Patches: 13341.patch + uploaded by putnopvut (license 60) (with small modification from + seanbright) + +2008-08-25 15:33 +0000 [r139764] Steve Murphy + + * main/pbx.c, res/res_features.c: This patch reverts the changes + made via 139347, and 139635, as users are seeing adverse + difference. I will un-close 13251. Back to the drawing board/ + concept/ beginning/ whatever! + +2008-08-22 22:24 +0000 [r139635] Steve Murphy + + * res/res_features.c: I found some problems with the code I + committed earlier, when I merged them into trunk, so I'm coming + back to clean up. And, in the process, I found an error in the + code I added to trunk and 1.6.x, that I'll fix using this patch + also. + +2008-08-22 21:36 +0000 [r139621] Jeff Peeler + + * main/manager.c: (closes issue #13359) Reported by: Laureano + Patches: originate_channel_check.patch uploaded by Laureano + (license 265) + +2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson + + * include/asterisk/threadstorage.h: Fix compilation when + DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported + by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy + (license 35) + + * main/frame.c: Remove show_frame_stats_deprecated since it is not + used anywhere and causes build errors if building under dev-mode + with TRACE_FRAMES selected in menuselect. (closes issue #13362) + Reported by: snuffy + + * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak! + + * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting + from incorrect locking order between iax2_pvt and ast_channel + structures. AST-13 + +2008-08-21 23:39 +0000 [r139387] Jeff Peeler + + * channels/chan_dahdi.c: Fixes loop that could possibly never exit + in the event of a channel never being able to be opened or + specify after a restart. (closes issue #11017) + +2008-08-21 23:03 +0000 [r139347] Steve Murphy + + * main/pbx.c, res/res_features.c: (closes issue #13251) Reported + by: sergee Tested by: murf THis is a bold move for a static + release fix, but I wouldn't have made it if I didn't feel + confident (at least a *bit* confident) that it wouldn't mess + everyone up. The reasoning goes something like this: 1. We simply + cannot do anything with CDR's at the current point (in pbx.c, + after the __ast_pbx_run loop). It's way too late to have any + affect on the CDRs. The CDR is already posted and gone, and the + remnants have been cleared. 2. I was very much afraid that moving + the running of the 'h' extension down into the bridge code (where + it would be now practical to do it), would result in a lot more + calls to the 'h' exten, so I implemented it as another exten + under another name, but found, to my pleasant surprise, that + there was a 1:1 correspondence to the running of the 'h' exten in + the pbx_run loop, and the new spot at the end of the bridge. So, + I ifdef'd out the current 'h' loop, and moved it into the bridge + code. The only difference I can see is the stuff about the + AST_PBX_KEEPALIVE, and hopefully, if this is still an important + decision point, I can replicate it if there are complaints. To be + perfectly honest, the KEEPALIVE situation is not totally clear to + me, and how it relates to a post-bridge situation is less clear. + I suspect the users will point out everything in total clarity if + this steps on anyone's toes! 3. I temporarily swap the bridge_cdr + into the channel before running the 'h' exten, which makes it + possible for users to edit the cdr before it goes out the door. + And, of course, with the endbeforehexten config var set, the + users can also get at the billsec/duration vals. After the h + exten finishes, the cdr is swapped back and processing continues + as normal. Please, all who deal with CDR's, please test this + version of Asterisk, and file bug reports as appropriate! + +2008-08-21 10:11 +0000 [r139283] Philippe Sultan + + * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4, + too. + +2008-08-20 22:14 +0000 [r139213] Russell Bryant + + * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The + issue here is that if you call ChanSpy and specify a spy group, + and sit in the application long enough looping through the + channel list, you will eventually run out of stack space and the + application with exit with a seg fault. The backtrace was always + inside of a harmless snprintf() call, so it was tricky to track + down. However, it turned out that the call to snprintf() was just + the biggest stack consumer in this code path, so it would always + be the first one to hit the boundary. (closes issue #13338) + Reported by: ruddy + +2008-08-20 19:52 +0000 [r139151] Shaun Ruffell + + * codecs/codec_dahdi.c: Fix bug where the samples were not accurate + when in G723 mode, which would cause the timestamp field of the + RTP header to be invalid. + +2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming + + * channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Backport support + for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not + doing so just makes it difficult for people with channels that + are in alarm when Asterisk starts up to get them going once the + alarm is cleared (closes issue #12160) Reported by: tzafrir + Patches: asterisk-chanalarms_14.patch uploaded by tzafrir + (license 46) Tested by: tzafrir + +2008-08-20 17:14 +0000 [r139074] Steve Murphy + + * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by: + murf The specialized reset routine is tromping on the flags field + of the CDR. I made a change to not reset the DISABLED bit. This + should get rid of this problem. + +2008-08-20 15:37 +0000 [r139015] Mark Michelson + + * channels/chan_sip.c: sip_read should properly handle a NULL + return from sip_rtp_read. (closes issue #13257) Reported by: + travishein + +2008-08-19 23:22 +0000 [r138949] Jeff Peeler + + * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL + compatability define for Zaptel + +2008-08-19 23:17 +0000 [r138942] Mark Michelson + + * channels/chan_agent.c: Reset agent_pvt variables back to the + values in agents.conf (from what the corresponding channel + variables were set to) when the agent logs out. (closes issue + #13098) Reported by: davidw Patches: + 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by + bbryant (license 36) Tested by: davidw + +2008-08-19 22:56 +0000 [r138938] Jeff Peeler + + * channels/chan_dahdi.c: Add configuration option to + chan_dahdi.conf to allow buffering policy and number of buffers + to be configured per channel. Syntax: buffers=, Where the number of buffers is some + non-negative integer and the policy is either "full", "half", or + "immediate". + +2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson + + * apps/app_chanspy.c: Add a lock and unlock prior to the + destruction of the chanspy_ds lock to ensure that no other + threads still have it locked. While this should not happen under + normal circumstances, it appears that if the spyer and spyee hang + up at nearly the same time, the following may occur. 1. + ast_channel_free is called on the spyee's channel. 2. The chanspy + datastore is removed from the spyee's channel in + ast_channel_free. 3. In the spyer's thread, the spyer attempts to + remove and destroy the datastore from the spyee channel, but the + datastore has already been removed in step 2, so the spyer + continues in the code. 4. The spyee's thread continues and calls + the datastore's destroy callback, chanspy_ds_destroy. This + involves locking the chanspy_ds. 5. Now the spyer attempts to + destroy the chanspy_ds lock. The problem is that in step 4, the + spyee has locked this lock, meaning that the spyer is attempting + to destroy a lock which is currently locked by another thread. + The backtrace provided in issue #12969 supports the idea that + this is possible (and has even occurred). This commit does not + close the issue, but should help in preventing one type of crash + associated with the use of app_chanspy. + + * apps/app_queue.c: Change the inequalities used in app_queue with + regards to timeouts from being strict to non-strict for more + accuracy. (closes issue #13239) Reported by: atis Patches: + app_queue_timeouts_v2.patch uploaded by atis (license 242) + +2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming + + * codecs/codec_dahdi.c: look for transcoder in proper place based + on build against Zaptel or DAHDI + +2008-08-18 11:57 +0000 [r138569] Sean Bright + + * channels/chan_dahdi.c: You know what's awesome? Code that + compiles... ;) + +2008-08-18 02:05 +0000 [r138516] Jeff Peeler + + * channels/chan_dahdi.c: fix compilation warnings + +2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler + + * channels/chan_dahdi.c: fixes use count to properly decrement if + an active dahdi channel is destroyed allowing module to be + unloaded + + * channels/chan_dahdi.c: add forgotten locks around ss_thread_count + in ss_thread for dahdi restart + +2008-08-15 22:33 +0000 [r138258] Tilghman Lesher + + * channels/chan_sip.c, configs/sip.conf.sample: More fixes for + realtime peers. (closes issue #12921) Reported by: Nuitari + Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 + (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 + (license 14) Tested by: Corydon76 + +2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler + + * channels/chan_dahdi.c: initialize condition variable + ss_thread_complete using ast_cond_init + + * channels/chan_dahdi.c: declared static mutexes using + AST_MUTEX_DEFINE_STATIC macro + + * channels/chan_dahdi.c: Fixes the dahdi restart functionality. + Dahdi restart allows one to restart all DAHDI channels, even if + they are currently in use. This is different from unloading and + then loading the module since unloading requires the use count to + be zero. Reloading the module is different in that the signalling + is not changed from what it was originally configured. Also, this + fixes not closing all the file descriptors for D-channels upon + module unload (which would prevent loading the module + afterwards). (closes issue #11017) + +2008-08-15 15:07 +0000 [r138027] Russell Bryant + + * main/autoservice.c: Ensure that when a hangup occurs in + autoservice, that a hangup frame gets properly deferred to be + read from the channel owner when it gets taken out of + autoservice. (closes issue #12874) Reported by: dimas Patches: + v1-12874.patch uploaded by dimas (license 88) + +2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher + + * funcs/func_strings.c: Additional check for more string specifiers + than arguments. (closes issue #13299) Reported by: adomjan + Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 + (license 14) func_strings.c-sprintf.patch uploaded by adomjan + (license 487) Tested by: adomjan + + * channels/chan_dahdi.c: Oops, wrong direction + + * channels/chan_dahdi.c: When creating the secondary subchannel + name, it is necessary to compare to the existing channel name + without the "Zap/" or "DAHDI/" prefix, since our test string is + also without that prefix. (closes issue #13027) Reported by: + dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer + (license 525) (Slightly modified by me, to compensate for both + names) + +2008-08-14 14:05 +0000 [r137731] Russell Bryant + + * configs/sip.conf.sample: Comments in this config file were + aligned only if your tab size was set to 8. So, convert tabs to + spaces so that things should be aligned regardless of what tab + size you use in your editor. + +2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming + + * Zaptel-to-DAHDI.txt: forgot one module name that changed + + * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c, + build_tools/menuselect-deps.in, configure, configure.ac, + codecs/codec_dahdi.c: add support for Zaptel versions that + contain the new transcoder interface + +2008-08-13 21:35 +0000 [r137580] Jeff Peeler + + * channels/chan_dahdi.c: Register DAHDISendKeypadFacility + application if dahdi_chan_mode is set to DAHDI + Zap. Mark + ZapSendKeypadFacility application as deprecated on usage. + +2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming + + * Zaptel-to-DAHDI.txt (added): add document describing what users + will need to be aware of when upgrading to this version and using + DAHDI + + * apps/app_meetme.c: remove some more chan_zap references + + * doc/asterisk-conf.txt, channels/chan_dahdi.c: document + dahdichanname option in doc/asterisk-conf.txt make chan_dahdi + read its configuration from zapata.conf if dahdichanname has been + set to 'no' + +2008-08-13 14:33 +0000 [r137348-137405] Sean Bright + + * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds + + * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends + and have it try to store CDR(userfield) if it is set. The new + behavior is to check for the userfield column on module load, and + if it exists, we will store CDR(userfield) when CDRs are written. + A similar patch already went into trunk and 1.6.0. (closes issue + #13290) Reported by: falves11 + +2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming + + * apps/app_meetme.c: convert this module to be able to handle DAHDI + or Zaptel (reported on asterisk-users, don't know how this got + missed before) + +2008-08-11 00:20 +0000 [r137138] Tilghman Lesher + + * res/res_odbc.c: Deallocate database connection handle on + disconnect, as we allocate another one on connect. (closes issue + #13271) Reported by: dveiga + +2008-08-09 17:11 +0000 [r136999] Russell Bryant + + * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate + to 0 if not found instead of empty. pointed out by tzafrir on + #asterisk-dev + +2008-08-09 15:25 +0000 [r136946] Tilghman Lesher + + * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged + revisions 136945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) + | 2 lines Regression fixes for Solaris ........ + +2008-08-08 00:15 +0000 [r136726] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c, + include/asterisk/ael_structs.h: (closes issue #13236) Reported + by: korihor Wow, this one was a challenge! I regrouped and ran a + new strategy for setting the ~~MACRO~~ value; I set it once per + extension, up near the top. It is only set if there is a switch + in the extension. So, I had to put in a chunk of code to detect a + switch in the pval tree. I moved the code to insert the set of + ~~exten~~ up to the beginning of the gen_prios routine, instead + of down in the switch code. I learned that I have to push the + detection of the switches down into the code, so everywhere I + create a new exten in gen_prios, I make sure to pass onto it the + values of the mother_exten first, and the exten next. I had to + add a couple fields to the exten struct to accomplish this, in + the ael_structs.h file. The checked field makes it so we don't + repeat the switch search if it's been done. I also updated the + regressions. + +2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming + + * build_tools/menuselect-deps.in, configure, configure.ac: change + the required dependency for codec_dahdi to only be satisfied by + DAHDI and not Zaptel, as the new transcoder interface is only in + DAHDI + +2008-08-07 18:14 +0000 [r136544] Shaun Ruffell + + * codecs/codec_dahdi.c: Updated codec_dahdi to use the new + transcoder interface in the first DAHDI release. Codec dahdi no + longer functions with the transcoder interface in zaptel at this + time (which the last zaptel release was 1.4.11). NOTE: Still + needs an update to the configure script to make sure that + codec_dahdi is only built if the new transcoder interface is + present in the drivers. (Issue: DAHDI-42) + +2008-08-07 16:50 +0000 [r136488] Tilghman Lesher + + * apps/app_queue.c: Update persistent state on all exit conditions. + (closes issue #12916) Reported by: sgenyuk Patches: + app_queue.patch.txt uploaded by neutrino88 (license 297) Tested + by: sgenyuk, aragon + +2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming + + * include/asterisk/doxyref.h: add a raw list of all libraries that + any part of Asterisk links directly to + + * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that + incorrectly ignores the casting away of 'const' for pointers when + the developer knows it is safe to do so + + * Makefile: remove config.cache during distclean, in case the user + is using autoconf caching + +2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher + + * channels/chan_dahdi.c: Also, parse + useincomingcalleridonzaptransfer (and add appropriate deprecation + warnings). + + * channels/chan_dahdi.c: For backwards compatibility with previous + 1.4 versions which used "zapchan" in users.conf, ensure that we + still support it. + +2008-08-06 21:18 +0000 [r136241] Richard Mudgett + + * channels/misdn_config.c, channels/chan_misdn.c, + configs/misdn.conf.sample: * The allowed_bearers setting in + misdn.conf misspelled one of its options: digital_restricted. * + Fixed some other spelling errors and typos. + +2008-08-06 20:42 +0000 [r136238] Mark Michelson + + * apps/app_queue.c: We only need to unregister the QueueStatus + manager command once on an unload + +2008-08-06 20:14 +0000 [r136190] Tilghman Lesher + + * contrib/init.d/rc.redhat.asterisk: -C option takes a filename, + not a directory path. (closes issue #13007) Reported by: + klaus3000 + +2008-08-06 18:58 +0000 [r136168] Russell Bryant + + * Makefile: Remove the use of --no-print-directory when compiling + subdirectories. This allows vim :make functionality to work + properly when errors have occurred in the build. Without printing + the directories, vim did not know how to find the file that the + error occurred in. If the extra bit of build noise annoys anyone, + just let me know, and I'll make this optional. + +2008-08-06 15:58 +0000 [r136062] Mark Michelson + + * main/rtp.c, channels/chan_skinny.c: Since adding the + AST_CONTROL_SRCUPDATE frame type, there are places where + ast_rtp_new_source may be called where the tech_pvt of a channel + may not yet have an rtp structure allocated. This caused a crash + in chan_skinny, which was fixed earlier, but now the same crash + has been reported against chan_h323 as well. It seems that the + best solution is to modify ast_rtp_new_source to not attempt to + set the marker bit if the rtp structure passed in is NULL. This + change to ast_rtp_new_source also allows the removal of what is + now a redundant pointer check from chan_skinny. (closes issue + #13247) Reported by: pj + +2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher + + * main/channel.c: Fix a longstanding bug in channel walking logic, + and fix the explanation to make sense. (Closes issue #13124) + + * main/translate.c: Since powerof() can return an error condition, + it's foolhardy not to detect and deal with that condition. + (Related to issue #13240) + + * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1) + Bugfix for debugging code 2) Reduce compiler warnings for another + section of debugging code (Closes issue #13237) + +2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson + + * /: Remove properties that should not be here + + * apps/app_skel.c: Revert inadvertent changes to app_skel that + occurred when I was testing for a memory leak + + * include/asterisk/abstract_jb.h, main/channel.c, /, + apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h: + Merging the issue11259 branch. The purpose of this branch was to + take into account "burps" which could cause jitterbuffers to + misbehave. One such example is if the L option to Dial() were + used to inject audio into a bridged conversation at regular + intervals. Since the audio here was not passed through the + jitterbuffer, it would cause a gap in the jitterbuffer's + timestamps which would cause a frames to be dropped for a brief + period. Now ast_generic_bridge will empty and reset the + jitterbuffer each time it is called. This causes injected audio + to be handled properly. ast_generic_bridge also will empty and + reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE + frame since the change in audio source could negatively affect + the jitterbuffer. All of this was made possible by adding a new + public API call to the abstract_jb called ast_jb_empty_and_reset. + (closes issue #11259) Reported by: plack Tested by: putnopvut + +2008-08-05 23:13 +0000 [r135799] Steve Murphy + + * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c, + include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit + Tested by: murf I discovered that also, in the previous bug fixes + and changes, the cdr.conf 'unanswered' option is not being + obeyed, so I fixed this. And, yes, there are two 'answer' times + involved in this scenario, and I would agree with you, that the + first answer time is the time that should appear in the CDR. (the + second 'answer' time is the time that the bridge was begun). I + made the necessary adjustments, recording the first answer time + into the peer cdr, and then using that to override the bridge + cdr's value. To get the 'unanswered' CDRs to appear, I purposely + output them, using the dial cmd to mark them as DIALED (with a + new flag), and outputting them if they bear that flag, and you + are in the right mode. I also corrected one small mention of the + Zap device to equally consider the dahdi device. I heavily tested + 10-sec-wait macros in dial, and without the macro call; I tested + hangups while the macro was running vs. letting the macro + complete and the bridge form. Looks OK. Removed all the + instrumentation and debug. + +2008-08-05 21:34 +0000 [r135747] Tilghman Lesher + + * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the + meaning of the flag IAX_HASCALLERID was perverted. This change + reverts IAX2 to the original meaning, which was, that the + callerid set on the client should be overridden on the server, + even if that means the resulting callerid is blank. In other + words, if you set "callerid=" in the IAX config, then the + callerid should be overridden to blank, even if set on the + client. Note that there's a distinction, even on realtime, + between the field not existing (NULL in databases) and the field + existing, but set to blank (override callerid to blank). + +2008-08-05 13:25 +0000 [r135597] Sean Bright + + * main/cli.c: Use PATH_MAX for filenames + +2008-08-04 20:15 +0000 [r135536] Russell Bryant + + * configs/chan_dahdi.conf.sample: fix a config sample typo + +2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher + + * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script + + * apps/app_voicemail.c: Memory leak on unload (closes issue #13231) + Reported by: eliel Patches: app_voicemail.leak.patch uploaded by + eliel (license 64) + +2008-08-04 16:26 +0000 [r135473] Russell Bryant + + * configs/chan_dahdi.conf.sample: Add a minor clarification to the + documentation of mohinterpret and mohsuggest + +2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak + + * apps/app_ices.c: make app_ices compile on OpenBSD. + + * channels/chan_skinny.c: fix some potential deadlocks in + chan_skinny (closes issue #13215) Reported by: qwell Patches: + 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) + Tested by: mvanbaak + +2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming + + * main/http.c: accomodate users who seem to lack a sense of humor + :-) + +2008-07-31 21:53 +0000 [r134976] Tilghman Lesher + + * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in + callfiles and manager, to allow video calls to be set up from + callfiles and AMI. (closes issue #9531) Reported by: Geisj + Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 + (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by + Corydon76 (license 14) Tested by: Corydon76 + +2008-07-31 19:37 +0000 [r134915] Russell Bryant + + * apps/app_ices.c: Get app_ices working again (closes issue #12981) + Reported by: dlogan Patches: + 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant + (license 36) 20080709__app_ices_v2_update_14.diff uploaded by + bbryant (license 36) Tested by: bbryant + +2008-07-31 19:23 +0000 [r134883] Steve Murphy + + * res/res_features.c: (closes issue #11849) Reported by: greyvoip + Tested by: murf OK, a few days of debugging, a bunch of + instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and + 5 solid notebook pages of notes later, I have made the small + tweek necc. to get the start time right on the second CDR when: A + Calls B B answ. A hits Xfer button on sip phone, A dials C and + hits the OK button, A hangs up C answers ringing phone B and C + converse B and/or C hangs up But does not harm the scenario + where: A Calls B B answ. B hits xfer button on sip phone, B dials + C and hits the OK button, B hangs up C answers ringing phone A + and C converse A and/or C hangs up The difference in start times + on the second CDR is because of a Masquerade on the B channel + when the xfer number is sent. It ends up replacing the CDR on the + B channel with a duplicate, which ends up getting tossed out. We + keep a pointer to the first CDR, and update *that* after the + bridge closes. But, only if the CDR has changed. I hope this + change is specific enough not to muck up any current CDR-based + apps. In my defence, I assert that the previous information was + wrong, and this change fixes it, and possibly other similar + scenarios. I wonder if I should be doing the same thing for the + channel, as I did for the peer, but I can't think of a scenario + this might affect. I leave it, then, as an exersize for the + users, to find the scenario where the chan's CDR changes and + loses the proper start time. + +2008-07-31 16:45 +0000 [r134814] Russell Bryant + + * channels/iax2-parser.c: In case we have some processing threads + that free more frames than they allocate, do not let the frame + cache grow forever. (closes issue #13160) Reported by: tavius + Tested by: tavius, russell + +2008-07-31 15:56 +0000 [r134758] Mark Michelson + + * apps/app_queue.c: Add more timeout checks into app_queue, + specifically targeting areas where an unknown and potentially + long time has just elapsed. Also added a check to try_calling() + to return early if the timeout has elapsed instead of potentially + setting a negative timeout for the call (thus making it have *no* + timeout at all). (closes issue #13186) Reported by: + miquel_cabrespina Patches: 13186.diff uploaded by putnopvut + (license 60) Tested by: miquel_cabrespina + +2008-07-30 22:39 +0000 [r134704] Tilghman Lesher + + * main/sched.c, include/asterisk/sched.h: Oops, wrong define + +2008-07-30 22:02 +0000 [r134652] Steve Murphy + + * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes + issue #13051) Reported by: pj This patch substitutes commas in + the expr supplied to the if () statement, as in if ( expr ) ... + This solves both the bugs above, and makes the source symmetric + with switch statements, which were earlier reported to need this + sort of treatment. I tested this using the examples, both for the + compiler and at run time. Looks good. + +2008-07-30 21:38 +0000 [r134649] Tilghman Lesher + + * configure, configure.ac: Qwell pointed out, via IRC, that the + previous fix only worked when explicitly set. When nothing is + set, and the option is implied, it breaks, because configure sets + the prefix to 'NONE'. Fixing. + +2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant + + * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max + stack size to fix a crash when compiled with LOW_MEMORY. (closes + issue #13154) Reported by: edantie + + * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread + that used this function leaked an allocation the size of a + pointer. (reported by jmls in #asterisk-dev) + +2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher + + * configure, configure.ac: Only override sysconfdir and mandir when + prefix=/usr (closes issue #13093) Reported by: pabelanger + + * res/res_agi.c: launch_netscript sometimes returns -1, which fails + to set AGISTATUS. Map failure to -1, so that AGISTATUS is always + set. (closes issue #13199) Reported by: smw1218 + +2008-07-30 18:31 +0000 [r134475] Mark Michelson + + * main/app.c: Fix a spot where a function could return without + bringing a channel out of autoservice. + +2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming + + * Makefile: use the proper method for building version.h + + * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c, + channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c, + apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c: + build against the now-typedef-free dahdi/user.h + +2008-07-29 15:54 +0000 [r134223] Mark Michelson + + * apps/app_voicemail.c: Merging the imap_consistency branch. The + main aim of this branch was to make the IMAP code function in the + same manner as the ODBC code does, eliminating the need for so + many IMAP-specific code chunks. The focal point of all of this + work was to make the various macros (e.g. RETRIEVE, DISPOSE) + functionally equivalent. While doing the above work, I also fixed + a few bugs that I came across in my testing. Among these were 1. + Fixed message forwarding. This was completely broken when using + IMAP. 2. Fixed the inability to save new messages as old and vice + versa. 3. Fixed the "delete" options in voicemail.conf when using + IMAP storage. Even though a few bugs were fixed and the code is a + lot more consistent, the one thing that was *not* improved in + this branch was performance. The merge of this to trunk may not + come immediately due to the amount of work it will probably + involve. (closes issue #12764) Reported by: balsamcn + +2008-07-28 21:50 +0000 [r134161] Tilghman Lesher + + * apps/app_voicemail.c: Detect when sox fails to raise the volume, + because sox can't read the file. (closes issue #12939) Reported + by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by + Corydon76 (license 14) Tested by: rickbradley + +2008-07-26 15:31 +0000 [r133980] Russell Bryant + + * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing + section to the docs in 1.4, as well, so that we can work on + having an accurate list for each version of Asterisk that is + supported + +2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher + + * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue + #13161) + + * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix + some errant device states by making the devicestate API more + strict in terms of the device argument (only without the unique + identifier appended). (closes issue #12771) Reported by: davidw + Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 + (license 14) Tested by: davidw, jvandal, murf + +2008-07-25 15:00 +0000 [r133578] Russell Bryant + + * /, LICENSE: Merged revisions 133577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) + | 2 lines Fix the IAX2 URI for calling Digium ........ + +2008-07-25 14:40 +0000 [r133572] Mark Michelson + + * channels/chan_sip.c: We need to make sure to null-terminate the + "name" portion of SIP URI parameters so that there are no bogus + comparisons. Thanks to bbryant for pointing this out. + +2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher + + * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes + issue #12707) + + * /: Blocked revisions 133360 via svnmerge ........ r133360 | + tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines + This part was not correctly patched for AST-2008-010. ........ + +2008-07-23 21:49 +0000 [r133295] Jason Parker + + * channels/chan_dahdi.c: inbandrelease is gone - it's now + inbanddisconnect + +2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming + + * include/asterisk/stringfields.h, main/utils.c: revert an + optimization that broke ABI... thanks russell! + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + apps/app_dahdibarge.c, channels/chan_dahdi.c, + apps/app_dahdiras.c: make some more changes to the dahdi/zap + channel name support stuff to ensure allthe globals are 'const', + and clean up mmichelson's changes to app_chanspy to simplify the + code + +2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + channels/chan_dahdi.c: As suggested by seanbright, the + PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at + compile time, since dahdi_chan_name is determined at load time. + Also changed the next_unique_id_to_use to have the static + qualifier. Also added the dahdi_chan_name_len variable so that + strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for + the suggestion. + + * apps/app_chanspy.c: Zap/pseudo is ten characters, but + DAHDI/pseudo is twelve. The strncmp call in next_channel should + account for this. + + * apps/app_chanspy.c: Update the "last" channel in next_channel in + app_chanspy so that the same pseudo channel isn't constantly + returned. related to issue #13124 + + * channels/chan_dahdi.c: Small cleanup. Move the declaration of the + DAHDI_SPANINFO variable to the block where it is used. This + allows one less #ifdef HAVE_PRI to clutter things up. Thanks to + Tzafrir for pointing this out on #asterisk-dev + + * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI + is not defined. + +2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming + + * channels/chan_dahdi.c: ensure that after a channel is created, if + it happened to be in 'channel alarm' state, when that alarm + clears we won't generate a spurious 'alarm cleared' message + (closes issue #12160) Reported by: tzafrir + + * include/asterisk/stringfields.h, main/utils.c: minor optimization + for stringfields: when a field is being set to a larger value + than it currently contains and it happens to be the most recent + field allocated from the currentl pool, it is possible to 'grow' + it without having to waste the space it is currently using (or + potentially even allocate a new pool) + +2008-07-23 11:37 +0000 [r132826] Christian Richter + + * channels/misdn/isdn_lib.c: another Fix because of r119585, this + commit has broken high frequented BRI Ports, there was a + possibility that a channel, that was marked as in_use would be + reused later, the corresponding port could got stuck then. So it + is recommended to upgrade for chan_misdn users. + +2008-07-22 22:14 +0000 [r132790] Mark Michelson + + * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly + within Asterisk. Prior to this change, a spiraled INVITE would + cause a 482 Loop Detected to be sent to the caller. With this + change, if a potential loop is detected, the Request-URI is + inspected to see if it has changed from what was originally + received. If pedantic mode is on, then this inspection is fully + RFC 3261 compliant. If pedantic mode is not on, then a string + comparison is used to test the equality of the two R-URIs. This + has been tested by using OpenSER to rewrite the R-URI and send + the INVITE back to Asterisk. (closes issue #7403) Reported by: + stephen_dredge + +2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming + + * include/asterisk/options.h, main/asterisk.c, + apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c, + apps/app_dahdiras.c: fix up namespace pollution for + dahdi_chan_mode enum correct registration of AMI actions in + chan_dahdi; in zap-only mode, only register the Zap flavors of + the actions (and use Zap prefixes for headers and acks), but in + dahdi+zap mode, register both Zap and DAHDI flavors of actions + + * Makefile.rules: add rules to create preprocessor output... useful + for debugging macros + +2008-07-22 21:19 +0000 [r132713] Tilghman Lesher + + * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged + revisions 132711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) + | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ + +2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming + + * channels/chan_dahdi.c: ensure that if any alarms exist at channel + creation time, they are handled identically to if they occurred + later, so that later alarm clearing will work properly and 'make + sense' (closes issue #12160) Reported by: tzafrir + + * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK + able to print a 'pretty' description of what it is doing + +2008-07-22 20:10 +0000 [r132645] Olle Johansson + + * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most + common question on the #asterisk iRC channel and on mailing lists + seems to be in regards to an error message when retransmit fails. + This is frequently misunderstood as a failure of Asterisk, not a + failure of the network to reach the other party. This document + tries to assist the Asterisk user in sorting out these issues by + explaining the logic and pointing at some possible causes. + Hopefully, we will get other questions now :-) + +2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming + + * channels/chan_dahdi.c: correct wording in comment + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac: use renamed + libpri API call for controlling this feature (was improperly + named before) + + * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel + on BRI spans, and don't attempt to use channel 24 as a D-channel + on spans of unexpected sizes + +2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant + + * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set + properly when it isn't supported on a channel (yet _another_ + useful patch by eliel). (issue #13081) Reported by: eliel + Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested + by: eliel + + * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels + not being removed when a call was rejected (from the calling box, + not the box that denied the registration). Related to revisions + 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently + tested 1.4 with a backport from those revisions, so I didn't see + this problem (oops). + +2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming + + * LICENSE: grant a license exception to allow distribution of + Asterisk binaries that use the UW IMAP Toolkit (which is licensed + under a non-GPL-compatible license) + +2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher + + * main/say.c: Fix for Taiwanese number syntax (closes issue #12319) + Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch + uploaded by CharlesWang (license 444) + + * main/config.c: Textual clarification (closes issue #13106) + Reported by: flefoll Patches: + config.c.br14.120173.patch-unknown-directive uploaded by flefoll + (license 244) + + * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within + the destroy, to allow a scheduled job to continue, if it's + waiting on the mutex which the destroy thread has. + + * main/sched.c: Oops + + * main/sched.c, include/asterisk/sched.h: Preserve ABI + compatibility with last change + + * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c: + Make the ast_assert call within ast_sched_del report something + useful. + +2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming + + * main/dlfcn.c (removed), main/loader.c, main/Makefile, + include/asterisk/dlfcn-compat.h (removed): remove the dlfcn + compatibility stuff, because no platforms that Asterisk currently + runs on it use it, and it doesn't build anyway + +2008-07-18 15:34 +0000 [r131915] Brett Bryant + + * res/res_features.c: Fix a bug in blind transfers where the + BLINDTRANSFER variable isn't always set to the other end of the + blind transfer. (closes issue #12586) + +2008-07-17 20:35 +0000 [r131790] Tilghman Lesher + + * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965) + as it appears that it was in error. This should fix talk call + progress on analog lines. (closes issue #12178) Reported by: + michael-fig Patches: 20080717__bug12178.diff.txt uploaded by + Corydon76 (license 14) + +2008-07-16 22:17 +0000 [r131491] Brett Bryant + + * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed + peers to register with case-insensitive names (user_cmp_cb and + peer_cmp_cb are now both case-sensitive). (closes issue #13091) + +2008-07-16 21:46 +0000 [r131480] Tilghman Lesher + + * channels/chan_iax2.c: Apparently, in certain cases, a callno is + already destroyed when iax2_destroy is called. + +2008-07-16 20:47 +0000 [r131421] Russell Bryant + + * channels/chan_iax2.c: Always ensure that the channel's tech_pvt + reference is NULL after calling the destroy callback. (closes + issue #13060) Reported by: jpgrayson Patches: + chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license + 492) + +2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson + + * apps/app_queue.c: Move the init_queue call back to where it used + to be (changed Sept 12 last year). It was moved then to prevent a + memory leak. Since then, the same memory leak recurred and was + fixed in a better way. Now it has been found that the placement + of this init_queue call can cause problems if a realtime queue + has values changed to an empty string. The problem is that the + default value for that queue parameter would not be set. (closes + issue #13084) Reported by: elbriga + + * apps/app_queue.c: Apparently, "thread safety" is important, + whatever that means. :P (Thanks Russell!) + + * apps/app_queue.c: Make absolutely certain that the transfer + datastore is removed from the calling channel once the caller is + finished in the queue. This could have weird con- sequences when + dialing local queue members when multiple transfers occur on a + single call. Also fixed a memory leak that would occur when an + attended transfer occurred from a queue member. (closes issue + #13047) Reported by: festr + +2008-07-16 17:53 +0000 [r131242] Steve Murphy + + * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The + problem was that, esoteric as it is, because the hangerupper + context immediately preceded the std-priv-extent macro, that the + checking code accidentally would fall from traversing hangerupper + into the std-priv-exten macro, where it would hit the hangerupper + in the 'includes', and proceed into an infinite recursion. A + small fix to traverse into the statements of the context instead + of the context solves this issue. I also added some commented out + printfs for debug, which were pretty handy in the face of a dorky + gdb. This was a problem around since the package was first + written; but evidently pretty rare in turning up in the field. + +2008-07-15 17:47 +0000 [r131012] Michiel van Baak + + * main/cdr.c: remove 4 lines of redundant code. (closes issue + #13080) Reported by: gknispel_proformatique Patches: + trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) + +2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher + + * main/manager.c, channels/chan_sip.c: astman_send_error does not + need a newline appended -- the API takes care of that for us. + (closes issue #13068) Reported by: gknispel_proformatique + Patches: asterisk_1_4_astman_send.patch uploaded by gknispel + (license 261) asterisk_trunk_astman_send.patch uploaded by + gknispel (license 261) + + * channels/chan_iax2.c: Override the callerid in all cases when the + callerid is set in the user, not just when a remote callerid is + set. Also, if not set in the user, allow the remote CallerID to + pass through. (closes issue #12875) Reported by: dimas Patches: + 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) + +2008-07-14 17:50 +0000 [r130792] Mark Michelson + + * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in + app_dial to be sure there are no audiohooks present on the + channels involved. This fixed a one-way audio situation I had in + my test setup. I couldn't find any open issues that suggested + one-way audio with regards to mixmonitor (or other audiohook) + usage, though. + +2008-07-14 17:10 +0000 [r130735] Michiel van Baak + + * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when + dnsmgr is not enabled. Previously this command would + automagically appear and disappear. This was confusing. (closes + issue #12796) Reported by: chappell Patches: + dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: + russell, chappell, mvanbaak + +2008-07-14 10:38 +0000 [r130634] Russell Bryant + + * main/audiohook.c: Bump up the debug level for a message. + +2008-07-13 22:48 +0000 [r130573] Michiel van Baak + + * main/manager.c: fix memory leak when originate from manager + cannot create a thread (closes issue #13069) Reported by: + gknispel_proformatique Patches: + asterisk_trunk_action_originate.patch uploaded by gknispel + (license 261) Tested by: gknispel_proformatique, mvanbaak + +2008-07-13 17:56 +0000 [r130514] Tilghman Lesher + + * channels/chan_iax2.c: Reverting 2 changesets, as it breaks + incoming IAX2 calls (Related to issue #12963) Reported by: + mvanbaak + +2008-07-12 10:25 +0000 [r130373] Michiel van Baak + + * pbx/pbx_ael.c: in 1.4 the functions still have | as argument + seperator. This commit fixes the use of RAND in the ael random + function. (closes issue #13061) Reported by: danpwi + +2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming + + * Makefile: forcibly remove the modules that are changing names + + * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c, + Makefile, main/channel.c, apps/app_dahdibarge.c, + channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c, + apps/app_dahdiras.c, main/file.c, + contrib/utils/zones2indications.c, include/asterisk/channel.h, + channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility + work, with lots more to come... this tree is not yet ready for + users to be easily upgrading or switching, but it needs to be :-) + +2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson + + * main/audiohook.c: Remove redundant logic + + * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While + this change has not been proven to fix any specific issue, it is + incorrect and could cause unforeseen problems. + +2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher + + * channels/chan_iax2.c: Ensure that a destination callno of 0 will + not match for frames that do not start a dialog (new, lagrq, and + ping). (closes issue #12963) Reported by: russellb Patches: + chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) + + * channels/chan_agent.c: Pass the devicestate from an underlying + channel up through the Agent channel. This should make the Agent + always report the correct device state, even when the underlying + channel is used for other purposes. (closes issue #12773) + Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded + by Corydon76 (license 14) Tested by: davidw + +2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming + + * doc/configuration.txt, configs/extensions.conf.sample, + configs/sla.conf.sample, configs/zapata.conf.sample (removed), + contrib/scripts/autosupport, README, + configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c, + include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt, + configs/extensions.ael.sample, configs/smdi.conf.sample: new + installations should be using DAHDI instead of Zaptel, so the + sample config file is now chan_dahdi.conf instead of zapata.conf + also, convert remaining references to zapata.conf in various + places + + * configs/zapata.conf.sample, channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: add support for a + configuration parameter for 'inband audio during RELEASE', which + is currently mandatory in libpri-1.4.4 but will become + configurable in libpri-1.4.5 later today (related to issue + #13042) + +2008-07-11 14:18 +0000 [r129970] Russell Bryant + + * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro + ... + +2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming + + * main/astmm.c: simplify calculation + + * main/astmm.c: fix a flaw found while experimenting with structure + alignment and padding; low-fence checking would not work properly + on 64-bit platforms, because the compiler was putting 4 bytes of + padding between the fence field and the allocation memory block + added a very obvious runtime warning if this condition reoccurs, + so the developer who broke it can be chastised into fixing it :-) + + * sounds/Makefile: don't attempt to set user/group ownership of + extracted sound files (reported on asterisk-users) (closes issue + #13059) + +2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher + + * channels/chan_iax2.c: Correctly deal with duplicate NEW frames + (due to retransmission). Also, fixup the destination call number + matching to be more strict and reliable. (closes issue #12963) + Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch + uploaded by jpgrayson (license 492) Tested by: jpgrayson, + Corydon76 + + * res/res_config_odbc.c: Oops + +2008-07-10 16:03 +0000 [r129567] Russell Bryant + + * sample.call: Note that pbx_spool.so is the module used for call + files (inspired by a question in #asterisk) + +2008-07-10 13:57 +0000 [r129505] Sean Bright + + * main/editline: Update svn:ignore + +2008-07-09 19:32 +0000 [r129436] Mark Michelson + + * main/rtp.c: Fix a problem where inbound rfc2833 audio would be + sent to the core instead of being P2P bridged. When the core + regenerated the rfc2833 packet for the outbound leg, the SSRC + would be different than the RTP audio on the call leg causing + DTMF detection issues on the far end. (closes issue #12955) + Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by + tsearle (license 373) Tested by: tonyredstone + +2008-07-09 13:41 +0000 [r129343] Sean Bright + + * main/editline/makelist (removed), main/editline/makelist.in + (added), main/editline/configure, main/editline/Makefile.in, + main/editline/configure.in: Look for the system installed awk + instead of assuming it's at /usr/bin/awk. Pointed out by jmls via + #asterisk-dev. + +2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson + + * doc/imapstorage.txt: Update documentation to have the correct + option name + + * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related + timeouts to IMAP voicemail in 1.4 since it should solve bugs + people are experiencing. Specifically, there are times where + communication with the IMAP server causes system calls to block + forever. If this should happen when querying the mailbox so that + chan_sip's do_monitor thread can send MWI to a phone, it means + that SIP calls cannot be processed any more. The timeout options + are outlined in doc/imapstorage.txt. Defaults for the timeouts + are sixty seconds. (closes issue #12987) Reported by: mthomasslo + +2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher + + * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h: + Cause SIP to return a 480 instead of a 404 when a sip peer + exists, but is not registered. (closes issue #12885) Reported by: + ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 + (license 14) Tested by: ibc + + * channels/chan_iax2.c: Timestamp decoding for video mini-frames is + bogus, because the timestamp only includes 15 bits, unlike voice + frames, which contain a 16-bit timestamp. (closes issue #13013) + Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch + uploaded by jpgrayson (license 492) + +2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson + + * channels/chan_sip.c: Don't hangup the call if we can't resolve + the Contact if there's a proxy route set for the call. ---- This + comment was added a while ago and today it hit me badly. /* OEJ: + Possible issue that may need a check: If we have a proxy route + between us and the device, should we care about resolving the + contact or should we just send it? */ + + * channels/chan_sip.c: Fix issues where repeated messages where + ignored, but retransmitted reliably instead of unreliably. + Reported by: johan Patches: 12746.txt uploaded by oej (license + 306) Tested by: johan (issue #12746) + +2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher + + * apps/app_voicemail.c: Check for non-NULL before stripping + characters. (closes issue #12954) Reported by: bfsworks Patches: + 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) + Tested by: deti + + * apps/app_voicemail.c: Stop using deprecated method, as requested + by Kevin. + +2008-07-07 22:41 +0000 [r128795] Russell Bryant + + * channels/chan_iax2.c: Fix handling of when a pvt disappears. + Properly return the pvt locked and don't hold the pvt lock while + destroying the ast_channel. (closes issue #13014) Reported by: + jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by + jpgrayson (license 492) + +2008-07-07 20:47 +0000 [r128737] Sean Bright + + * channels/chan_iax2.c: Remove spurious trailing whitespace from + log messages and fix a spelling error in a log message. (closes + issue #13017) Reported by: jpgrayson Patches: + chan_iax2_space_after_newline.patch uploaded by jpgrayson + (license 492) chan_iax2_spelling.patch uploaded by jpgrayson + (license 492) + +2008-07-07 17:02 +0000 [r128639] Mark Michelson + + * channels/chan_iax2.c: By using the iaxdynamicthreadcount to + identify a thread, it was possible for thread identifiers to be + duplicated. By using a globally-unique monotonically- increasing + integer, this is now avoided. (closes issue #13009) Reported by: + jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by + jpgrayson (license 492) + +2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming + + * configure, configure.ac: use tzafrir's patch to fix this problem + properly... i made the previous set of changes without thoroughly + testing them, doh! (closes issue #12911) Reported by: tzafrir + Patches: custum_dahdi_configure_2.diff uploaded by tzafrir + (license 46) Tested by: tzafrir + +2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher + + * pbx/pbx_config.c: Move the free down one + + * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the + 'dialplan remove extension' logic, so that it a) works with + cidmatch, and b) completes contexts correctly when the extension + is ambiguous. (closes issue #12980) Reported by: licedey Patches: + 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76 + +2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming + + * apps/Makefile: remove this, it has been moved to the main + Makefile + + * Makefile, main/editline/np/vis.c: a couple of small + Solaris-related fixes (closes issue #11885) Reported by: snuffy, + asgaroth + + * configure, main/Makefile, configure.ac, acinclude.m4: ensure that + DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places + needed improve AST_EXT_LIB_CHECK to accept (and remember) + additional CFLAGS data like it does in trunk already (closes + issue #12911) Reported by: tzafrir + +2008-07-03 00:16 +0000 [r127663] Steve Murphy + + * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, + channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h: + The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported + by: murf Tested by: murf, deeperror (closes issue #12907) + Reported by: falves11 Tested by: murf, falves11 (closes issue + #11849) Reported by: greyvoip As to 11849, I think these changes + fix the core problems brought up in that bug, but perhaps not the + more global problems created by the limitations of CDR's + themselves not being oriented around transfers. Reopen if necc, + but bug reports are not the best medium for enhancement + discussions. We need to start a second-generation CDR + standardization effort to cover transfers. (closes issue #11093) + Reported by: rossbeer Tested by: greyvoip, murf + +2008-07-02 20:47 +0000 [r127560] Mark Michelson + + * channels/chan_agent.c: Fix thread-safety of some of the + pbx_builtin_getvar_helper calls + +2008-07-02 19:47 +0000 [r127501] Tilghman Lesher + + * main/acl.c: Merged revisions 127466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | + tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines + Solaris fix (closes issue #12949) Reported by: snuffy Patches: + bug_12949.diff uploaded by snuffy (license 35) ........ + +2008-07-01 23:36 +0000 [r127244] Mark Michelson + + * apps/app_voicemail.c: Add error message to failed open(2) calls + inside the copy() function of app_voicemail. This idea came as + part of my work in helping to resolve issue #12764. + +2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher + + * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old, + slow search for matching callno in chan_iax2 (but allow it to be + reenabled for debugging) + + * channels/chan_iax2.c: Oops + + * channels/chan_iax2.c: Change around how we schedule pings and + lagrqs, and fix a reason why the jobs were not getting properly + cancelled. (closes issue #12903) Reported by: stevedavies + Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 + (license 14) Tested by: stevedavies + + * channels/chan_iax2.c: Suppress annoying warning by finding the + remaining cases where the callno is not in the hash. + +2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson + + * channels/chan_sip.c: Use domain part of SIP uri in register= + configuration as fromdomain. Reported by: one47 Patches: + sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes + issue #12474) + + * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch + by oej and IgorG. Reported by: IgorG Patches: + bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by: + IgorG, oej (closes issue #12299) + + * configs/sip.conf.sample: Clear up documentation on "domain=" + setting in sip.conf Reported by: davidw (closes issue #12413) + + * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs + requests (to confirm that the dialog exist). Don't bother + checking the request URI. (closes issue #11264) Reported by: ibc + + * channels/chan_sip.c: Fix bad XML for hold notification. Reported + by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432) + (closes issue #12942) + +2008-06-30 23:11 +0000 [r126680] Jeff Peeler + + * channels/chan_dahdi.c: Load the proper channel configuration file + based on which driver was detected. + +2008-06-30 22:30 +0000 [r126674] Tilghman Lesher + + * configs/zapata.conf.sample: Add note about other names for + EuroISDN + +2008-06-30 16:05 +0000 [r126573] Russell Bryant + + * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS + version of the recently added DEADLOCK_AVOIDANCE() macro. This + caused the lock to not actually be released, and as a result, not + avoid deadlocks at all. This resolves the issues reported in the + last while about Asterisk locking up all over the place (and most + commonly, in chan_iax2). (closes issue #12927) (closes issue + #12940) (closes issue #12925) (potentially closes others ...) + +2008-06-30 12:50 +0000 [r126516] Olle Johansson + + * channels/chan_sip.c: Send all responses to an INVITE reliably, so + that we retransmit if we don't get an ACK and also fail if we + don't get the very same precious ACK. Based on patch by tsearle, + with my own additions. (closes issue #12951) Reported by: tsearle + Patches: busy_retransmit.patch uploaded by tsearle (license 373) + +2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming + + * pbx/Makefile: ignore warnings for prototypes in GTK headers + +2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher + + * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying + to re-register. (closes issue #12863) Reported by: ricvil + + * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined + to HAVE_ZAPTEL in dahdi_compat.h, we must first check for + HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches: + tonezone_compat.diff uploaded by opticron (license 267) + + * main/utils.c, include/asterisk/lock.h: In this debugging + function, copy to a buffer instead of using potentially unsafe + pointers. + + * channels/chan_local.c: Add proper deadlock avoidance. (closes + issue #12914) Reported by: ozan Patches: + 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14) + Tested by: ozan + +2008-06-26 23:03 +0000 [r125587] Jason Parker + + * main/utils.c: Make sure to unlock the lock_info lock (huh?). + Possible deadlock? + +2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson + + * apps/app_queue.c: Add the interface of a queue member to the + output of the "queue show" command so that it can easily be + associated with a queue member's name. This helps so that the + appropriate queue member can be removed or paused since the + interface is required, not the member's name. (closes issue + #12783) Reported by: davevg Patches: app_queue.diff uploaded by + davevg (license 209) with small mod from me + + * apps/app_queue.c: Backport of attended transfer queue_log patch + from trunk. This patch allows for attended transfers to be logged + in the queue_log the same way that blind transfers have always + been. It was decided by popular opinion on the asterisk-dev + mailing list that this should be backported to 1.4. Thanks to + everyone who gave an opinion. + + * apps/app_queue.c: Prior to this patch, the "queue show" command + used cached information for realtime queues instead of giving + up-to-date info. Now realtime is queried for the latest and + greatest in queue info. (closes issue #12858) Reported by: bcnit + Patches: queue_show.patch uploaded by putnopvut (license 60) + +2008-06-26 16:32 +0000 [r125384] Olle Johansson + + * channels/chan_sip.c: Add support for peer realm based auth (a few + missing lines, the rest is well documented but never worked) + +2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming + + * channels/chan_dahdi.c: ensure that (whenever possible) if we + generate a log message because an ioctl() call to DAHDI/Zaptel + failed, that we include the reason it failed by including the + stringified error number (issue AST-80) + +2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher + + * main/rtp.c: Check for rtcp structure before trying to delete + schedule. (closes issue #12872) Reported by: destiny6628 Patches: + 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) + Tested by: destiny6628 + + * configs/agents.conf.sample: Document ackcall=always. (closes + issue #12852) Reported by: davidw + +2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming + + * apps/app_rpt.c, include/asterisk/dahdi_compat.h, + channels/chan_dahdi.c, configure, + include/asterisk/tonezone_compat.h (added), configure.ac: allow + tonezone to live in a different place than DAHDI/Zaptel, since + dahdi-tools and dahdi-linux are now separate packages and can be + installed in different places don't include tonezone.h in + dahdi_compat.h, because only a couple of modules need it get + app_rpt building again after the DAHDI changes (closes issue + #12911) Reported by: tzafrir + +2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher + + * channels/chan_dahdi.c: Pvt deadlock causes some channels to get + stuck in Reserved status. (closes issue #12621) Reported by: + fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by + Corydon76 (license 14) Tested by: fabianoheringer + + * apps/app_voicemail.c: Occasionally control characters find their + way into CallerID. These need to be stripped prior to placing + CallerID in the headers of an email. (closes issue #12759) + Reported by: RobH Patches: 20080602__bug12759__2.diff.txt + uploaded by Corydon76 (license 14) Tested by: RobH + + * channels/chan_sip.c: Don't access the pvt structure if unable to + acquire the lock. (closes issue #12162) Reported by: norman + Patches: 12162-lockfail.diff uploaded by qwell (license 4) + +2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming + + * channels/chan_iax2.c: emit a warning if the old IAX2 call + searching code finds a call when the new code did not... so that + we can get rid of the old code in 2-3 months + +2008-06-22 02:54 +0000 [r124540] Steve Murphy + + * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac + Sorry, my testing did not contain the simple case of forkCDR(v), + I am much embarrassed to admit. If I had, I would have more + solidly initialized the opts element for varset. + +2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher + + * apps/app_rpt.c: usleep with a value over 1,000,000 is + nonportable. Changing to use sleep() instead. (closes issue + #12814) Reported by: pputman Patches: app_rtp_sleep.patch + uploaded by pputman (license 81) + + * main/app.c: If the last character in a string to be parsed is the + delimiter, then we should count that final empty string as an + additional argument. + +2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach + + * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in + digium-mib.txt and asterisk-mib.txt to placate smilint - bug + 12905 + +2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher + + * channels/chan_local.c: When using a Local channel, started by a + call file, with a destination of an AGI script, the AGI script + does not always get notified of a hangup if the underlying + channel hangs up early. (closes issue #11833) Reported by: IgorG + Patches: local_hangup-v1.diff uploaded by IgorG (license 20) + + * channels/chan_dahdi.c: It's possible for a hangup to be received, + even just after the initial cid spill. (closes issue #12453) + Reported by: Alex728 Patches: 20080604__bug12453.diff.txt + uploaded by Corydon76 (license 14) + +2008-06-19 20:28 +0000 [r124112] Mark Michelson + + * apps/app_voicemail.c: Fix IMAP forwarding so that messages are + sent to the proper mailbox. (closes issue #12897) Reported by: + jaroth Patches: destination_forward.patch uploaded by jaroth + (license 50) + +2008-06-19 19:55 +0000 [r124066] Brett Bryant + + * main/utils.c: Merge revision 124064 from trunk. Add errors that + report any locks held by threads when they are being closed. + +2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher + + * main/channel.c: Change informative messages to use the _multiple + variant when multiple formats are possible. (Closes issue #12848) + Reported by klaus3000 + + * build_tools/strip_nonapi: Only process 40 arguments (20 files) at + once with xargs, because some older shells may force xargs to + separate on an odd boundary. (Closes issue #12883) Reported by + Nik Soggia + + * configs/sip.conf.sample: Correct description of notifyringing + option. (Closes issue #12890) Reported by gminet + + * main/asterisk.c: The RDTSC instruction was introduced on the + Pentium line of microprocessors, and is not compatible with + certain 586 clones, like Cyrix. Hence, asking for i386 + compatibility was always incorrect. See + http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported + by tecnoxarxa + + * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add + support for saying numbers in Hebrew. (closes issue #11662) + Reported by: greenfieldtech Patches: say.c.patch-12042008 + uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods + uploaded by greenfieldtech (with signficant changes to the + spreadsheet by me) + + * pbx/pbx_spool.c: Set the variables top-down, so that if a script + sets a variable more than once, the last one will take + precedence. (closes issue #12673) Reported by: phber Patches: + 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) + +2008-06-17 20:26 +0000 [r123485] Mark Michelson + + * channels/chan_sip.c: Make chan_sip build under dev mode with + compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this + +2008-06-17 18:56 +0000 [r123391] Tilghman Lesher + + * channels/chan_iax2.c: Fix 3 more places where not locking the + structure could cause the wrong lock to be unlocked. (Closes + issue #12795) + +2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson + + * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab + between the Cseq number and SIP request method in the Cseq: + header. Asterisk did not handle this properly, but with this + patch, all is well. (closes issue #12834) Reported by: tobias_e + Patches: 12834.patch uploaded by putnopvut (license 60) Tested + by: tobias_e + + * apps/app_queue.c: davidw pointed out that the holdtime + calculation used by app_queue does not use "boxcar" filtering as + the comments say. The term "boxcar" means that the number of + samples used to calculate stays constant, with new samples + replacing the oldest ones. The queue holdtime calculation uses + all holdtime samples collected since the queue was loaded, so the + comment has been changed to be accurate. (closes issue #12781) + Reported by: davidw + +2008-06-17 15:48 +0000 [r123271] Russell Bryant + + * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out + by seanbright. When a container got destroyed, the underlying + bucket list entry for each object that was in the container at + that time did not get free'd. + +2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher + + * channels/chan_mgcp.c, channels/chan_dahdi.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_iax2.c: Port "hasvoicemail" change from SIP to + other channel drivers + + * channels/chan_sip.c: People expect that if "hasvoicemail" is set + in users.conf, even if "mailbox" isn't set, that SIP will detect + a mailbox. (closes issue #12855) Reported by: PLL Patches: + 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: PLL + +2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp + + * channels/chan_sip.c: Only compare the first 15 characters so that + even if the charset is specified we still accept it as SDP. + (closes issue #12803) Reported by: lanzaandrea Patches: + chan_sip.c.diff uploaded by lanzaandrea (license 496) + + * channels/chan_sip.c: Don't send a BYE on a dialog that is already + gone during a REFER. (closes issue #12865) Reported by: flefoll + Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by + flefoll (license 244) + +2008-06-13 21:44 +0000 [r122713] Mark Michelson + + * main/autoservice.c: Short circuit the loop in autoservice_run if + there are no channels to poll. If we continued, then the result + would be calling poll() with a NULL pollfd array. While this is + fine with POSIX's poll(2) system call, those who use Asterisk's + internal poll mechanism (Darwin systems) would have a failed + assertion occur when poll is called. (related to issue #10342) + +2008-06-13 18:57 +0000 [r122663] Jeff Peeler + + * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed + dahdi compatability header from assuming either dahdi or zaptel + is installed (may not have either) + +2008-06-13 17:45 +0000 [r122617] Terry Wilson + + * apps/app_dial.c: Remove extra option from previous solution + attempt + +2008-06-13 17:36 +0000 [r122613] Jeff Peeler + + * configure, configure.ac: (closes issue #12846) Reported by: + Netview Tested by: jpeeler Use correct location to search for + tonezone. + +2008-06-13 16:29 +0000 [r122589] Terry Wilson + + * apps/app_dial.c, res/res_features.c: This should fix the behavior + of the 'T' dial feature being passed incorrectly to the + transferee when builtin_atxfers are used. Also, doing a + builtin_atxfer to parking was broken and is fixed here as well. + (closes issue #11898) Reported by: sergee Tested by: otherwiseguy + +2008-06-12 19:08 +0000 [r122314] Jeff Peeler + + * main/indications.c, include/asterisk/dahdi_compat.h (added), + main/loader.c, main/channel.c, channels/chan_dahdi.c (added), + configure, apps/app_zapscan.c (removed), apps/app_zapras.c + (removed), main/app.c, include/asterisk/options.h, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c, + channels/chan_zap.c (removed), apps/app_page.c, + include/asterisk/indications.h, apps/app_dahdiras.c (added), + configure.ac, apps/app_disa.c, include/asterisk/channel.h, + apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c + (removed), channels/chan_misdn.c, apps/app_flash.c, + build_tools/menuselect-deps.in, funcs/func_channel.c, + main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c, + codecs/codec_dahdi.c (added), res/res_indications.c, + pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c, + main/asterisk.c, apps/app_dahdibarge.c (added), + apps/app_meetme.c, include/asterisk/autoconfig.h.in, + apps/app_dahdiscan.c (added), acinclude.m4, + res/res_musiconhold.c, codecs/codec_zap.c (removed), + channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI + usage favored, but all Zap stuff should continue working. Release + announcement to follow. + +2008-06-12 18:50 +0000 [r122311] Mark Michelson + + * apps/app_queue.c: Properly play a holdtime message if the + announce-holdtime option is set to "once." (closes issue #12842) + Reported by: ramonpeek Patches: patch001.diff uploaded by + ramonpeek (license 266) + +2008-06-12 18:22 +0000 [r122259] Russell Bryant + + * channels/chan_iax2.c: Fix some race conditions that cause + ast_assert() to report that chan_iax2 tried to remove an entry + that wasn't in the scheduler + +2008-06-12 15:46 +0000 [r122208] Jeff Peeler + + * apps/app_parkandannounce.c, res/res_features.c: (closes issue + #12193) Reported by: davidw Patch by: Corydon76, modified by me + to work properly with ParkAndAnnounce app + +2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher + + * apps/app_meetme.c: Flipflop the sections for two options, since + the section for 'X' (exit context) may otherwise absorb + keypresses meant for 's' (admin/user menu). (closes issue #12836) + Reported by: blitzrage Patches: 20080611__bug12836.diff.txt + uploaded by Corydon76 (license 14) Tested by: blitzrage + + * main/channel.c: Occasionally, the alertpipe loses its nonblocking + status, so detect and correct that situation before it causes a + deadlock. (Reported and tested by ctooley via #asterisk-dev) + +2008-06-12 14:51 +0000 [r122127] Steve Murphy + + * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the + code added to ast_cdr_busy, _failed, and _noanswer was redundant. + Didn't spot it until I was resolving conflicts in trunk. Ugh. + Redundant code removed. It wasn't harmful. Just dumb. + +2008-06-12 Russell Bryant + + * Asterisk 1.4.21 released. + +2008-06-06 Russell Bryant + + * Asterisk 1.4.21-rc2 released. + +2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant + + * UPGRADE-1.2.txt: fix filename + + * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file + from Asterisk 1.2, for handy reference. + +2008-06-05 16:56 +0000 [r120675] Philippe Sultan + + * res/res_jabber.c: Ignore appended resource when comparing JIDs. + +2008-06-05 16:38 +0000 [r120671] Russell Bryant + + * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the + forwarding station isn't very useful for most people, so pull in + the changes that allow searching for SMDI messages based on other + components of the SMDI message. Also, update the SMDI + documentation. + +2008-06-04 22:05 +0000 [r120513] Mark Michelson + + * apps/app_queue.c: Make sure that the string we set will survive + the unref of the queue member. Thanks to Russell, who pointed + this out. + +2008-06-04 18:35 +0000 [r120425] Tilghman Lesher + + * channels/chan_zap.c: If we fail to setup the PRI request channel, + don't continue, exit with an error. (closes issue #11989) + Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt + uploaded by Corydon76 (license 14) + +2008-06-04 16:26 +0000 [r120371] Russell Bryant + + * pbx/pbx_config.c: Make the "dialplan remove include" CLI command + actually work. Also, tweak some formatting, and make the success + message a little bit more clear. (closes AST-52) + +2008-06-04 14:11 +0000 [r120285] Mark Michelson + + * apps/app_queue.c: Tab completion when removing a member should + give the member's interface, not the name, since the interface is + what is expected for the command. (closes issue #12783) Reported + by: davevg + +2008-06-04 13:31 +0000 [r120282] Joshua Colp + + * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message + for when the dialplan is done reloading. (closes issue #12716) + Reported by: chappell Patches: dialplan_reload_2.diff uploaded by + chappell (license 8) + +2008-06-03 22:41 +0000 [r120226] Tilghman Lesher + + * pbx/pbx_loopback.c: Due to incorrect use of the + AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform + any translation on the extension number before searching for it + in the target context. (closes issue #12473) Reported by: + chappell Patches: pbx_loopback.c.diff uploaded by chappell + (license 8) + +2008-06-03 22:15 +0000 [r120173] Jeff Peeler + + * main/config.c: (closes issue #11594) Reported by: yem Tested by: + yem This change decreases the buffer size allocated on the stack + substantially in config_text_file_load when LOW_MEMORY is turned + on. This change combined with the fix from revision 117462 + (making mkintf not copy the zt_chan_conf structure) was enough to + prevent the crash. + +2008-06-03 21:34 +0000 [r120168] Russell Bryant + + * channels/chan_iax2.c: Fix another place where peer->callno could + change at a very bad time, and also fix a place where a peer was + used after the reference was released. (inspired by rev 120001) + +2008-06-03 Russell Bryant + + * Asterisk 1.4.21-rc1 released. + +2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher + + * main/manager.c: When listing the manager users, managers in + users.conf are not shown, even though they are allowed to + connect. (closes issue #12594) Reported by: bkruse Patches: + 12594-managerusers-2.diff uploaded by qwell (license 4) Tested + by: bkruse + + * channels/chan_iax2.c: Save the callno when we're poking, because + our peer structure could change during deadlock avoidance (and + thus we unlock the wrong callno, causing a cascade failure). + (closes issue #12717) Reported by: gewfie Patches: + 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) + Tested by: gewfie + +2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL. + Hadn't updated this for the changes I made to preserve ${EXTEN} + in switches, which affected several tests because it adds extra + priorities, and at least one needed to be updated because of the + removal of the empty extension warning message. + + * pbx/pbx_ael.c: as per + http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, + which is a message from Philipp Kempgen, requesting that the + WARNING that an extension is empty be reduced to a NOTICE or + less, as empty extensions are syntactically possible, and no big + deal. With which I agree, and have removed that WARNING message + entirely. I think it is not necessary to see this message. It + didn't state that a NoOp() was inserted automatically on your + behalf, and really, as users, who cares? Why freak out dialplan + writers with unnecessary warnings? The details of the + machinations a compiler goes thru to produce working assembly + code is of little interest to most programmers-- we will follow + the unix principal of doing our work silently. + +2008-06-03 14:46 +0000 [r119926] Joshua Colp + + * channels/chan_sip.c: Treat ECONNREFUSED as an error that will + stop further retransmissions. (issue #AST-58, patch from + Switchvox) + +2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant + + * channels/chan_iax2.c: Revert a change made for issue #12479. This + change caused a regression such that a dial string such as + (IAX2/foo) did not automatically fall back to dialing the 's' + extension anymore. (closes issue #12770) Reported by: dagmoller + + * main/manager.c: Improve CLI command blacklist checking for the + command manager action. Previously, it did not handle case or + whitespace properly. This made it possible for blacklisted + commands to get executed anyway. (closes issue #12765) + +2008-06-02 14:32 +0000 [r119740] Philippe Sultan + + * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest + account with any configured XMPP client (in jabber.conf). The + actual connection is made when a call comes in Asterisk. Fix the + ast_aji_get_client function that was not able to retrieve an XMPP + client from its JID. (closes issue #12085) Reported by: junky + Tested by: phsultan + +2008-06-02 12:30 +0000 [r119687] Russell Bryant + + * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get + sent because it comes up too soon, make sure to reschedule so it + gets sent later. + +2008-06-02 09:29 +0000 [r119585-119636] Christian Richter + + * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is + enabled + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added + counter for unhandled_bmsg Print, this prevents the logs to be + flooded to fast and save CPU in this error scenario. Added + 'last_used' element to bc structure, when a bchannel changes from + used to free this exact time will be marked in last_used. When a + new channel is requested the find_free_chan function will check + if the new empty channel was used within the last second, if yes + it will search for the next channel, if no it will return this + channel. This simple mechanism has prooven to prevent race + conditions where the NT and TE tried to allocate the exact same + channel at the same time (RELEASE cause: 44). + +2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant + + * channels/chan_iax2.c: Change a debug message to an actual debug + message + + * apps/app_dial.c: Fix another typo in documentation + +2008-06-01 20:47 +0000 [r119478] Michiel van Baak + + * apps/app_dial.c: small typo fix 'retires' => 'retries' + +2008-05-30 21:17 +0000 [r119404] Tilghman Lesher + + * apps/app_queue.c: When joinempty=strict, it only failed on join + if there were busy members. If all members were logged out OR + paused, then it (incorrectly) let callers join the queue. (closes + issue #12451) Reported by: davidw + +2008-05-30 19:46 +0000 [r119354] Joshua Colp + + * main/autoservice.c: Fix a bug I found while testing for another + issue. + +2008-05-30 16:44 +0000 [r119301] Michiel van Baak + + * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, + contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.slackware.asterisk: dont use a bashism way to + check the $VERSION variable. The rc/init.d scripts, and + safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2 + (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski) + (closes issue #12687) Reported by: loloski Patches: + 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by + mvanbaak (license 7) Tested by: loloski, mvanbaak + +2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 + May 2008) | 7 lines - Instead of only enforcing destination call + number checking on an ACK, check all full frames except for PING + and LAGRQ, which may be sent by older versions too quickly to + contain the destination call number. (As suggested by Tim Panton + on the asterisk-dev list) - Merge changes from + team/russell/iax2-frame-race, which prevents PING and LAGRQ from + being sent before the destination call number is known. ........ + + * main/autoservice.c: Fix a race condition in channel autoservice. + There was still a small window of opportunity for a DTMF frame, + or some other deferred frame type, to come in and get dropped. + (closes issue #12656) (closes issue #12656) Reported by: dimas + Patches: v3-12656.patch uploaded by dimas (license 88) -- with + some modifications by me + + * include/asterisk/audiohook.h: Oddly enough, all of the contents + of audiohook.h were in there twice. I have removed the second + copy. + +2008-05-29 20:24 +0000 [r119071] Tilghman Lesher + + * channels/chan_zap.c: Call waiting tone occurs too often, because + it's getting serviced by both subchannels. (closes issue #11354) + Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded + by Corydon76 (license 14) + +2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant + + * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones - + use ast_safe_sleep() instead of calling the wait application + (thanks to tilghman for pointing these out!) + + * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 + May 2008) | 7 lines Merge changes from + team/russell/iax2-another-fix-to-the-fix As described in the + following post to the asterisk-dev mailing list, only enforce + destination call numbers when processing an ACK. + http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html + (closes issue #12631) ........ + + * apps/app_milliwatt.c: - Mark app_milliwatt dependent on + res_indications (thanks to jsmith) - fix a typo in a log message + (thanks to qwell) + + * apps/app_milliwatt.c: Change milliwatt to use the proper tone by + default (1004 Hz) instead of 1000 Hz. An option is there to use + 1000 Hz for anyone that might want it. + +2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher + + * include/asterisk/lock.h: Define also when not DEBUG_THREADS + + * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c, + channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, + include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging + code that ensures that when we do deadlock avoidance, we don't + lose the information about how a lock was originally acquired. + +2008-05-29 00:25 +0000 [r118858] Steve Murphy + + * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes + issue #11721) (closes issue #12726) Reported by: arkadia Tested + by: murf These changes: 1. revert the changes made via bug 10668; + I should have known that such changes, even tho they made sense + at the time, seemed like an omission, etc, were actually integral + to the CDR system via forkCDR. It makes sense to me now that + forkCDR didn't natively end any CDR's, but rather depended on + natively closing them all at hangup time via traversing and + closing them all, whether locked or not. I still don't completely + understand the benefits of setvar and answer operating on locked + cdrs, but I've seen enough to revert those changes also, and stop + messing up users who depended on that behavior. bug 12726 found + reverting the changes fixed his changes, and after a long review + and working on forkCDR, I can see why. 2. Apply the suggested + enhancements proposed in 10668, but in a completely compatible + way. ForkCDR will behave exactly as before, but now has new + options that will allow some actions to be taken that will + slightly modify the outcome and side-effects of forkCDR. Based on + conversations I've had with various people, these small tweaks + will allow some users to get the behavior they need. For + instance, users executing forkCDR in an AGI script will find the + answer time set, and DISPOSITION set, a situation not covered + when the routines were first written. 3. A small problem in the + cdr serializer would output answer and end times even when they + were not set. This is now fixed. + +2008-05-28 16:10 +0000 [r118716] Brett Bryant + + * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 -- + Fixes a bug in chan_iax that uses send_command to poke a peer + while a channel is unlocked in some cases, and because it can + cause seemingly random failures could be related to some bugs in + the tracker... + +2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an + option to use the source IP address of RTP as the destination IP + address of UDPTL when a specific option is enabled. If the remote + side is properly configured (ports forwarded) then UDPTL will + flow. (closes issue #10417) Reported by: cstadlmann + + * channels/chan_sip.c: Fix an issue where codec preferences were + not set on dialogs that were not authenticated via a user or peer + and allow framing to work without rtpmap in the SDP. (closes + issue #12501) Reported by: slimey + +2008-05-27 19:15 +0000 [r118551] Tilghman Lesher + + * main/cli.c: When showing an error message for a command, don't + shorten the command output, as it tends to confuse the user (it's + fine for suggesting other commands, however). Reported by: + seanbright (on #asterisk-dev) Fixed by: me + +2008-05-27 19:07 +0000 [r118509] Mark Michelson + + * apps/app_chanspy.c: Russell noted to me that in the case that + separate threads use their own addressing system, the fix I made + for issue 12376 does not guarantee uniqueness to the datastores' + uids. Though I know of no system that works this way, I am going + to change this right now to prevent trying to track down some + future bug that may occur and cause untold hours of debugging + time to track down. The change involves using a global counter + which increases with each new chanspy_ds which is created. This + guarantees uniqueness. + +2008-05-27 18:58 +0000 [r118465] Tilghman Lesher + + * main/asterisk.c: NULL character should terminate only commands + back to the core, not log messages to the console. (closes issue + #12731) Reported by: seanbright Patches: + 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14) + Tested by: seanbright + +2008-05-27 17:17 +0000 [r118416] Michiel van Baak + + * apps/app_voicemail.c: small update to the g() option of + app_voicemail to note that gain changes only work on zap channels + right now. issue #12578 shows it's not clear right now. + +2008-05-27 16:38 +0000 [r118365] Mark Michelson + + * apps/app_chanspy.c: Add a unique id to the datastore allocated in + app_chanspy since it is possible that multiple spies may be + listening to the same channel. (closes issue #12376) Reported by: + DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut + (license 60) Tested by: destiny6628 (closes issue #12243) + Reported by: atis + +2008-05-27 15:45 +0000 [r118358] Tilghman Lesher + + * configs/queues.conf.sample: Add a note that pbx_config.so is + needed for Local channels. (Closes issue #12671) + +2008-05-25 16:02 +0000 [r118251] Tilghman Lesher + + * channels/chan_sip.c: Realtime flag affects construction in + multiple ways, so consulting whether rtcachefriends was set was + done too soon (needed to be done inside build_peer, not just as a + flag to build_peer). Also, fullcontact needed to be + reconstructed, because realtime separates the embedded ';' into + multiple fields. (closes issue #12722) Reported by: barthpbx + Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 + (license 14) Tested by: barthpbx (Much of the discussion happened + on #asterisk-dev for diagnosing this issue) + +2008-05-23 21:21 +0000 [r118163] Jeff Peeler + + * channels/chan_zap.c: Fix a few things I missed to ensure + zt_chan_conf structure is not modified in mkintf + +2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher + + * include/asterisk/utils.h: Add format type checking for recently + de-inlined function + + * doc/cli.txt (added), doc/00README.1st: Add information on using + the Asterisk console, including tab command line completion. + (Closes issue #12681) + +2008-05-23 12:30 +0000 [r118048] Russell Bryant + + * include/asterisk/utils.h, main/utils.c: Don't declare a function + that takes variable arguments as inline, because it's not valid, + and on some compilers, will emit a warning. + http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes + issue #12289) Reported by: francesco_r Patches by Tilghman, final + patch by me + +2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher + + * main/asterisk.c: Also remove preamble from asynchronous events + (reported by jsmith on #asterisk-dev) + + * funcs/func_realtime.c: Take into account the length of delimiters + when calculating result string length. (closes issue #12696) + Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch + uploaded by adomjan (license 487) + +2008-05-21 20:11 +0000 [r117582] Jeff Peeler + + * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure + is not modified in mkintf. + +2008-05-21 19:38 +0000 [r117574] Joshua Colp + + * channels/chan_sip.c: Apply the autoframing setting to dialogs + that do not get matched against a user or peer. + +2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher + + * pbx/pbx_spool.c: Revert accidental commit of the last change + + * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the + output also when -rx is not being used (Related to issue #12702) + +2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant + + * main/asterisk.c: Don't filter the magic character in the network + verboser. It gets filtered once it reaches the client. (related + to issue #12702, pointed out by tilghman) + + * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose + marker in front of every message from ast_verbose() being sent to + remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose + marker. (related to issue #12702) + + * main/asterisk.c: Don't display the verbose marker for calls to + ast_verbose() that do not include a VERBOSE_PREFIX in front of + the message. (closes issue #12702) Reported by: johnlange Patched + by me + +2008-05-21 16:58 +0000 [r117462] Jeff Peeler + + * channels/chan_zap.c: Pass a pointer for the conf parameter to the + function mkintf rather than the whole zt_chan_conf structure. + +2008-05-20 Russell Bryant + + * Asterisk 1.4.20 released. + +2008-05-14 Russell Bryant + + * Asterisk 1.4.20-rc3 released. + +2008-05-14 12:51 +0000 [r116230] Olle Johansson + + * channels/chan_sip.c: Accept text messages even with Content-Type: + text/plain;charset=Södermanländska + +2008-05-13 23:47 +0000 [r116088] Mark Michelson + + * main/channel.c, include/asterisk/lock.h: A change to the way + channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. + After debugging a deadlock, it was noticed that when + DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin + of channel locks is obscured by the fact that all channel locks + appear to happen in the function ast_channel_lock(). This code + change redefines ast_channel_lock to be a macro which maps to + __ast_channel_lock(), which then relays the proper file name, + line number, and function name information to the core lock + functions so that this information will be displayed in the case + that there is some sort of locking error or core show locks is + issued. + +2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant + + * channels/chan_local.c: Fix a deadlock involving channel + autoservice and chan_local that was debugged and fixed by + mmichelson and me. We observed a system that had a bunch of + threads stuck in ast_autoservice_stop(). The reason these threads + were waiting around is because this function waits to ensure that + the channel list in the autoservice thread gets rebuilt before + the stop() function returns. However, the autoservice thread was + also locked, so the autoservice channel list was never getting + rebuilt. The autoservice thread was stuck waiting for the channel + lock on a local channel. However, the local channel was locked by + a thread that was stuck in the autoservice stop function. It + turned out that the issue came down to the local_queue_frame() + function in chan_local. This function assumed that one of the + channels passed in as an argument was locked when called. + However, that was not always the case. There were multiple cases + in which this channel was not locked when the function was + called. We fixed up chan_local to indicate to this function + whether this channel was locked or not. The previous assumption + had caused local_queue_frame() to improperly return with the + channel locked, where it would then never get unlocked. (closes + issue #12584) (related to issue #12603) + + * main/autoservice.c: Fix an issue that I noticed in autoservice + while mmichelson and I were debugging a different problem. I + noticed that it was theoretically possible for two threads to + attempt to start the autoservice thread at the same time. This + change makes the process of starting the autoservice thread, + thread-safe. + +2008-05-13 20:28 +0000 [r115944] Joshua Colp + + * channels/chan_alsa.c: Use the right flag to open the audio in + non-blocking. (closes issue #12616) Reported by: + nicklewisdigiumuser + +2008-05-13 18:36 +0000 [r115884] Tilghman Lesher + + * main/asterisk.c: If the socket dies (read returns 0=EOF), return + immediately. (Closes issue #12637) + +2008-05-12 17:51 +0000 [r115735] Mark Michelson + + * main/utils.c: If a thread holds no locks, do not print any + information on the thread when issuing a core show locks command. + This will help to de-clutter output somewhat. Russell said it + would be fine to place this improvement in the 1.4 branch, so + that's why it's going here too. + +2008-05-09 16:34 +0000 [r115579] Joshua Colp + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Improve res_ninit and res_ndestroy autoconf logic on the Darwin + platform. + +2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant + + * channels/chan_iax2.c: Remove debug output. + + * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 + May 2008) | 25 lines Fix a race condition that bbryant just found + while doing some IAX2 testing. He was running Asterisk trunk + running IAX2 calls through a few Asterisk boxes, however, the + audio was extremely choppy. We looked at a packet trace and saw a + storm of INVAL and VNAK frames being sent from one box to + another. It turned out that what had happened was that one box + tried to send a CONTROL frame before the 3 way handshake had + completed. So, that frame did not include the destination call + number, because it didn't have it yet. Part of our recent work + for security issues included an additional check to ensure that + frames that are supposed to include the destination call number + have the correct one. This caused the frame to be rejected with + an INVAL. The frame would get retransmitted for forever, rejected + every time ... This race condition exists in all versions that + got the security changes, in theory. However, it is really only + likely that this would cause a problem in Asterisk trunk. There + was a control frame being sent (SRCUPDATE) at the _very_ + beginning of the call, which does not exist in 1.2 or 1.4. + However, I am fixing all versions that could potentially be + affected by the introduced race condition. These changes are what + bbryant and I came up with to fix the issue. Instead of simply + dropping control frames that get sent before the handshake is + complete, the code attempts to wait a little while, since in most + cases, the handshake will complete very quickly. If it doesn't + complete after yielding for a little while, then the frame gets + dropped. ........ + + * channels/chan_sip.c: Don't give up on attempting an outbound + registration if we receive a 408 Timeout. (closes issue #12323) + + * contrib/scripts/postgres_cdr.sql (removed): remove + postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as + well (closes issue #9676) + + * contrib/init.d/rc.debian.asterisk: Don't exit the script if + Asterisk is not running. (closes issue #12611) + + * main/pbx.c: Don't use a channel before checking for channel + allocation failure. (closes issue #12609) Reported by: edantie + + * contrib/init.d/rc.debian.asterisk: Use the same method for + executing Asterisk as the rest of the script. (closes issue + #12611) Reported by: b_plessis + +2008-05-07 Russell Bryant + + * Asterisk 1.4.20-rc2 released. + +2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant + + * channels/chan_sip.c: Track peer references when stored in the + sip_pvt struct as the peer related to a qualify ping or a + subscription. This fixes some realtime related crashes. (closes + issue #12588) (closes issue #12555) + +2008-05-06 19:55 +0000 [r115418-115422] Jason Parker + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | + 7 lines read requires an argument on some non-bash shells (closes + issue #12593) Reported by: bkruse Patches: + getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) + ........ + + * res/res_musiconhold.c: Switch to using ast_random() rather than + just rand(). This does not fix the bug reported, but I believe it + is correct. (from issue #12446) Patches: bug_12446.diff uploaded + by snuffy (license 35) + +2008-05-06 19:31 +0000 [r115415] Tilghman Lesher + + * main/asterisk.c: Don't print the terminating NUL. (Closes issue + #12589) + +2008-05-06 13:54 +0000 [r115341] Joshua Colp + + * configure, configure.ac: Add in missing argument. + +2008-05-05 22:50 +0000 [r115333] Tilghman Lesher + + * main/asterisk.c, main/logger.c: Separate verbose output from CLI + output, by using a preamble. (closes issue #12402) Reported by: + Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt + uploaded by Corydon76 (license 14) + 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by + Corydon76 (license 14) + +2008-05-05 22:10 +0000 [r115327] Joshua Colp + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, codecs/codec_speex.c, + configure.ac: Make sure that either the main speex library + contains preprocess functions or that speexdsp does. If both fail + then speex stuff can not be built. + +2008-05-05 21:41 +0000 [r115320] Mark Michelson + + * apps/app_queue.c: Don't consider a caller "handled" until the + caller is bridged with a queue member. There was too much of an + opportunity for the member to hang up (either during a delay, + announcement, or overly long agi) between the time that he + answered the phone and the time when he actually was bridged with + the caller. The consequence of this was that if the member hung + up in that interval, then proper abandonment details would not be + noted in the queue log if the caller were to hang up at any point + after the member hangup. (closes issue #12561) Reported by: + ablackthorn + +2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher + + * Makefile: Reverse order, such that user configs override default + selections + + * include/asterisk/res_odbc.h: Err, the documentation on the return + value of ast_odbc_backslash_is_escape is exactly backwards. + +2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant + + * channels/chan_sip.c: Avoid putting opaque="" in Digest + authentication. This patch came from switchvox. It fixes + authentication with Primus in Canada, and has been in use for a + very long time without causing problems with any other providers. + (closes issue AST-36) + +2008-05-05 03:22 +0000 [r115285] Tilghman Lesher + + * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, + contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug + out if Asterisk is already running. (closes issue #12525) + Reported by: explidous Patches: 20080428__bug12525.diff.txt + uploaded by Corydon76 (license 14) Tested by: mvanbaak + +2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp + + * configure, acinclude.m4: Expand the test function for GCC + attributes so that more complex attributes are properly + recognized. + + * include/asterisk/compiler.h: For my next trick I will make these + work with what our autoconf header file gives us. + + * configure, acinclude.m4: Treat warnings as errors when checking + if a GCC attribute exists. We have to do this as GCC will just + ignore the attribute and pop up a warning, it won't actually fail + to compile. + +2008-05-02 20:25 +0000 [r115257] Brett Bryant + + * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, CHANGES: Add new "pri show version" command to show + the libpri version for support reasons. + +2008-05-02 14:28 +0000 [r115196] Mark Michelson + + * include/asterisk/sched.h: Clarify a comment that was, well, just + wrong. It turns out that ignoring the way that macros expand. + Instead, I have clarified in the comment why the macro will work + even if the scheduler id for the task to be deleted changes + during the execution of the macro. + +2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher + + * include/asterisk/res_odbc.h: Change the comment of deprecated to + an actual compiler deprecation + + * main/utils.c: '#' is another reserved character for URIs that + also needs to be escaped. (closes issue #10543) Reported by: + blitzrage Patches: 20080418__bug10543.diff.txt uploaded by + Corydon76 (license 14) + +2008-05-01 Russell Bryant + + * Asterisk 1.4.20-rc1 released. + +2008-04-30 16:30 +0000 [r114891] Russell Bryant + + * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c: + Merge changes from team/russell/iax2_find_callno and + iax2_find_callno_1.4 These changes address a critical performance + issue introduced in the latest release. The fix for the latest + security issue included a change that made Asterisk randomly + choose call numbers to make them more difficult to guess by + attackers. However, due to some inefficient (this is by far, an + understatement) code, when Asterisk chose high call numbers, + chan_iax2 became unusable after just a small number of calls. On + a small embedded platform, it would not be able to handle a + single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run + more than about 16 IAX2 channels. Ouch. These changes address + some performance issues of the find_callno() function that have + bothered me for a very long time. On every incoming media frame, + it iterated through every possible call number trying to find a + matching active call. This involved a mutex lock and unlock for + each call number checked. So, if the random call number chosen + was 20000, then every media frame would cause 20000 locks and + unlocks. Previously, this problem was not as obvious since + Asterisk always chose the lowest call number it could. A second + container for IAX2 pvt structs has been added. It is an astobj2 + hash table. When we know the remote side's call number, the pvt + goes into the hash table with a hash value of the remote side's + call number. Then, lookups for incoming media frames are a very + fast hash lookup instead of an absolutely insane array traversal. + In a quick test, I was able to get more than 3600% more IAX2 + channels on my machine with these changes. + +2008-04-30 16:23 +0000 [r114890] Olle Johansson + + * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created + in Huntsville together with Mark M (putnopvut) (closes issue + #12363) Reported by: jvandal Tested by: putnopvut, oej + +2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming + + * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro + for indexing through the iaxs/iaxsl arrays so that the size of + the arrays can be adjusted in one place, and change the size of + the arrays from 32768 calls to 2048 calls when LOW_MEMORY is + defined + + * Makefile.rules: pay attention to *all* header files for + dependency tracking, not just the local ones (inspired by r578 of + asterisk-addons by tilghman) + +2008-04-29 19:40 +0000 [r114848] Mark Michelson + + * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel + variables instead of the channel's macrocontext and macroexten + fields. This is needed because if macros are daisy-chained, the + incorrect context and extension are placed on the new channel. I + also added locking to the channel prior to accessing these + variables as noted in trunk's janitor project file. (closes issue + #12549) Reported by: darren1713 Patches: + app_queue.c.macroextenpatch uploaded by darren1713 (license 116) + (with modifications from me) Tested by: putnopvut + +2008-04-29 17:08 +0000 [r114829] Jason Parker + + * res/res_config_pgsql.c: Change warning message to debug, since + there are cases where 0 results is perfectly fine. + +2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr + 2008) | 2 lines stop script from appending source code if run + multiple times ........ + +2008-04-28 04:47 +0000 [r114708] Tilghman Lesher + + * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are + embedded, they take on a different name, without the ".so" + extension. Specifically check for this name, when we're checking + if a module is loaded. (Closes issue #12534) + +2008-04-27 01:26 +0000 [r114695] Sean Bright + + * configure, configure.ac: When we don't explicitly pass a path to + the --with-tds configure option, we may end up finding tds.h in + /usr/local/include instead of /usr/include. If this happens, the + grep that looks for the version (from tdsver.h) will fail and + we'll have some problems during the build. + +2008-04-26 13:15 +0000 [r114689] Tilghman Lesher + + * contrib/scripts/vmail.cgi: Clicking forward without selecting a + message leaves an errant .lock file. (closes issue #12528) + Reported by: pukepail Patches: patch.diff uploaded by pukepail + (license 431) + +2008-04-25 21:54 +0000 [r114673] Russell Bryant + + * channels/chan_iax2.c: Use consistent logic for checking to see if + a call number has been chosen yet. Also, remove some redundant + logic I recently added in a fix. + +2008-04-25 19:32 +0000 [r114662] Mark Michelson + + * apps/app_chanspy.c: Move the unlock of the spyee channel to + outside the start_spying() function so that the channel is not + unlocked twice when using whisper mode. + +2008-04-25 15:53 +0000 [r114649] Tilghman Lesher + + * configs/zapata.conf.sample, configs/iax.conf.sample, + configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference + documentation files that actually exist. (closes issue #12516) + Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded + by linuxmaniac (license 472) + +2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson + + * channels/chan_sip.c: Re-invite RTP during a masquerade so that, + for instance, an AMI redirect of two channels which are natively + bridged will preserve audio on both channels. This prevents a + problem with Asterisk not re-inviting due to one of the channels + having being a zombie. (closes issue #12513) Reported by: + mneuhauser Patches: + asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by + mneuhauser (license 425) + + * apps/app_queue.c: Output of channel variables when + eventwhencalled=vars was set was being truncated two characters. + This patch corrects the problem. (closes issue #12493) Reported + by: davidw + + * channels/chan_local.c: Resolve a deadlock in chan_local by + releasing the channel lock temporarily. (closes issue #11712) + Reported by: callguy Patches: 11712.patch uploaded by putnopvut + (license 60) Tested by: acunningham + +2008-04-24 19:53 +0000 [r114621] Tilghman Lesher + + * channels/chan_local.c: Ensure that when we set the accountcode, + it actually shows up in the CDR. (Fix for AMI Originate) (Closes + issue #12007) + +2008-04-24 15:55 +0000 [r114608] Russell Bryant + + * channels/chan_iax2.c: Fix a silly mistake in a change I made + yesterday that caused chan_iax2 to blow up very quickly. (issue + #12515) + +2008-04-24 14:55 +0000 [r114603] Olle Johansson + + * channels/chan_sip.c: Only have one max-forwards header in + outbound REFERs. Discovered in the Asterisk SIP Masterclass in + Orlando. Thanks Joe! + +2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant + + * main/http.c: Improve some broken cookie parsing code. Previously, + manager login over HTTP would only work if the mansession_id + cookie was first. Now, the code builds a list of all of the + cookies in the Cookie header. This fixes a problem observed by + users of the Asterisk GUI. (closes AST-20) + + * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting + the correct next channel to not always work. Also, remove setting + the amount of time to wait for a digit from 5 seconds back down + to 1/10 of a second. I believe this was so the beep didn't get + played over and over really fast, but a while back I put in + another fix for that issue. (closes issue #12498) Reported by: + jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by + jsmith (license 15) + +2008-04-23 18:28 +0000 [r114594] Jason Parker + + * res/res_musiconhold.c: Fix reload/unload for res_musiconhold + module. (closes issue #11575) Reported by: sunder Patches: + M11575_14_rev3.diff uploaded by junky (license 177) + bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) + +2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant + + * main/manager.c, include/asterisk/manager.h: Store the manager + session ID explicitly as 4 byte ID instead of a ulong. The + mansession_id cookie is coded to be limited to 8 characters of + hex, and this could break logins from 64-bit machines in some + cases. (inspired by AST-20) + + * channels/chan_iax2.c: Fix find_callno_locked() to actually return + the callno locked in some more cases. + +2008-04-23 16:51 +0000 [r114584] Olle Johansson + + * channels/chan_sip.c: Add 502 support for both directions, not + only one... (see r114571) + +2008-04-23 14:54 +0000 [r114579] Joshua Colp + + * main/pbx.c: Instead of stopping dialplan execution when SayNumber + attempts to say a large number that it can not print out a + message informing the user and continue on. (closes issue #12502) + Reported by: bcnit + +2008-04-22 23:51 +0000 [r114571] Tilghman Lesher + + * channels/chan_sip.c: Treat a 502 just like a 503, when it comes + to processing a response code + +2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant + + * channels/chan_iax2.c: When we receive a full frame that is + supposed to contain our call number, ensure that it has the + correct one. (closes issue #10078) (AST-2008-006) + + * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I + thought I was going to be able to leave 1.4 alone, but that was + not the case. I ran into some problems with G.722 in 1.4, so I + have merged in all of the fixes in this area that I have made in + trunk/1.6.0, and things are happy again. + + * res/res_musiconhold.c: Trivial change to read the number of + samples from a frame before calling ast_write() + + * res/res_features.c: After a parked call times out, allow the call + back to the parker to time out. (closes issue #10890) + + * channels/chan_iax2.c: If the dial string passed to the call + channel callback does not indicate an extension, then consider + the extension on the channel before falling back to the default. + (closes issue #12479) Reported by: darren1713 Patches: + exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license + 116) + + * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from + team/russell/issue_9520 These changes make sure that the + reference count for sip_peer objects properly reflects the fact + that the peer is sitting in the scheduler for a scheduled + callback for qualifying peers or for expiring registrations. + Without this, it was possible for these callbacks to happen at + the same time that the peer was being destroyed. This was + especially likely to happen with realtime peers, and for people + making use of the realtime prune CLI command. (closes issue + #9520) Reported by: kryptolus Committed patch by me + +2008-04-21 14:39 +0000 [r114322] Joshua Colp + + * channels/chan_sip.c: Only drop audio if we receive it without a + progress indication. We allow other frames through such as DTMF + because they may be needed to complete the call. (closes issue + #12440) Reported by: aragon + +2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher + + * apps/app_playback.c: Ensure that help text terminates with a + newline + + * res/res_musiconhold.c: MOH usage information needs a terminating + newline, or else "asterisk -rx 'help moh reload'" will hang. + Reported via -dev list, fixed by me. + +2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant + + * main/manager.c: Don't destroy a manager session if poll() returns + an error of EAGAIN. + + * Makefile: ensure directories are created before we try to install + stuff into them + + * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for + bininstall + +2008-04-18 17:44 +0000 [r114257] Mark Michelson + + * channels/chan_zap.c, main/callerid.c: Clearing up error messages + so they make a bit more sense. Also removing a redundant error + message. Issue AST-15 + +2008-04-18 15:24 +0000 [r114248] Russell Bryant + + * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL) + (closes issue #12476) Reported by: davidw Patch by me + +2008-04-18 13:33 +0000 [r114245] Sean Bright + + * channels/chan_sip.c: Only complete the SIP channel name once for + 'sip show channel ' + +2008-04-18 06:49 +0000 [r114242] Tilghman Lesher + + * apps/app_setcallerid.c: For consistency sake, ensure that the + values that ${CALLINGPRES} returns are valid as an input to + SetCallingPres. (Closes issue #12472) + +2008-04-17 22:15 +0000 [r114230] Russell Bryant + + * main/autoservice.c: Remove redundant safety net. The check for + the autoservice channel list state accomplishes the same goal in + a better way. (issue #12470) Reported By: atis + +2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson + + * apps/app_chanspy.c: Declaration of the peer channel in this scope + was making it so the peer variable defined in the outer scope was + never set properly, therefore making iterating through the + channel list always restart from the beginning. This bug would + have affected anyone who called chanspy without specifying a + first argument. (closes issue #12461) Reported by: stever28 + + * main/frame.c, include/asterisk/dsp.h: Add prototype for + ast_dsp_frame_freed. I'm not sure how this was compiling + before... + + * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was + possible for a reference to a frame which was part of a freed DSP + to still be referenced, leading to memory corruption and eventual + crashes. This code change ensures that the dsp is freed when we + are finished with the frame. This change is very similar to a + change Russell made with translators back a month or so ago. + (closes issue #11999) Reported by: destiny6628 Patches: + 11999.patch uploaded by putnopvut (license 60) Tested by: + destiny6628, victoryure + +2008-04-17 16:23 +0000 [r114204] Russell Bryant + + * Makefile: Fix the bininstall target to install from subdirs, as + well. (closes issue AST-8, patch from bmd at switchvox) + +2008-04-17 13:42 +0000 [r114198] Philippe Sultan + + * res/res_jabber.c: Use keepalives effectively in order diagnose + bug #12432. + +2008-04-17 12:56 +0000 [r114195] Tilghman Lesher + + * res/res_agi.c: Add special case for when the agi cannot be + executed, to comply with the documentation that we return failure + in that case. (closes issue #12462) Reported by: fmueller + Patches: 20080416__bug12462.diff.txt uploaded by Corydon76 + (license 14) Tested by: fmueller + +2008-04-17 10:51 +0000 [r114191] Sean Bright + + * apps/app_chanspy.c: Make sure we have enough room for the + recording's filename. + +2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming + + * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by + initializing the structure to all zeroes in case it contains + fields that we don't write values into (which it does as of + Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian + +2008-04-16 19:59 +0000 [r114180] Tilghman Lesher + + * channels/chan_vpb.cc: Backport revisions for latest vpb drivers + to 1.4 (Closes issue #12457) + +2008-04-16 17:30 +0000 [r114173] Jason Parker + + * channels/chan_zap.c: Fix "fallthrough" behavior here, so config + options in a previously configured user don't override settings + in general. (closes issue #12458) Reported by: tzafrir Patches: + chanzap_users_sections.diff uploaded by tzafrir (license 46) + +2008-04-16 14:10 +0000 [r114167] Joshua Colp + + * apps/app_meetme.c: Include the proper headers for using mkdir on + FreeBSD. (closes issue #12430) Reported by: ys Patches: + app_meetme.c.diff uploaded by ys (license 281) + +2008-04-15 20:26 +0000 [r114148] Olle Johansson + + * channels/chan_sip.c: Handle subscribe queues in all situations... + Thanks to festr_ on irc for telling me about this bug. + +2008-04-15 17:17 +0000 [r114120-114138] Jason Parker + + * contrib/scripts/autosupport: Update Digium autosupport script, + for more useful information. (closes issue #12452) Reported by: + angler Patches: autosupport.diff uploaded by angler (license 106) + + * apps/app_queue.c: Allow autofill to work in the general section + of queues.conf. Additionally, don't try to (re)set options when + they have empty values in realtime (all unset columns would have + an empty value). (closes issue #12445) Reported by: atis Patches: + 12445-autofill.diff uploaded by qwell (license 4) + + * channels/chan_h323.c: The call_token on the pvt can occasionally + be NULL, causing a crash. If it is NULL, we can skip this + channel, since it can't the one we're looking for. (closes issue + #9299) Reported by: vazir + +2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson + + * main/channel.c: Increase the retry count when attempting to show + channels. This apparently cleared an issue someone was seeing + when attempting to show channels when the load was high. (closes + issue #11667) Reported by: falves11 Patches: 11677.txt uploaded + by russell (license 2) Tested by: falves11 + + * apps/app_dial.c, apps/app_queue.c: If the datastore has been + moved to another channel due to a masquerade, then freeing the + datastore here causes an eventual double free when the new + channel hangs up. We should only free the datastore if we were + able to successfully remove it from the channel we are + referencing (i.e. the datastore was not moved). (closes issue + #12359) Reported by: pguido + + * main/channel.c: Save a local copy of the generate callback prior + to unlocking the channel in case the generate callback goes NULL + on us after the channel is unlocked. Thanks to Russell for + pointing this need out to me. + +2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp + + * channels/chan_sip.c: It is possible for the remote side to say + they want T38 but not give any capabilities. (closes issue + #12414) Reported by: MVF + + * main/rtp.c: Don't change the SSRC when a new source comes into + play, this might happen quite often and depending on the remote + side... they might not like this. (closes issue #12353) Reported + by: dimas + +2008-04-11 22:32 +0000 [r114083] Terry Wilson + + * channels/chan_iax2.c: Several places in the code called + find_callno() (which releases the lock on the pvt structure) and + then immediately locked the call and did things with it. + Unfortunately, the call can disappear between the find_callno and + the lock, causing Bad Stuff(tm) to happen. Added + find_callno_locked() function to return the callno withtout + unlocking for instances that it is needed. (issue #12400) + Reported by: ztel + +2008-04-11 21:35 +0000 [r114072] Jason Parker + + * main/pbx.c: It's possible that a channel can have an async goto + on the successful execution of an application as well. Closes + issue #12172. + +2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson + + * res/res_features.c: Fix a race condition that may happen between + a sip hangup and a "core show channel" command. This patch adds + locking to prevent the resulting crash. (closes issue #12155) + Reported by: tsearle Patches: show_channels_crash2.patch uploaded + by tsearle (license 373) Tested by: tsearle + + * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when + LOW_MEMORY is enabled. + + * channels/chan_sip.c: Be sure that we're not about to set + bridgepvt NULL prior to dereferencing it. (closes issue #11775) + Reported by: fujin + +2008-04-10 17:26 +0000 [r114035] Jason Parker + + * main/file.c: Only try to prefix language if we are not using an + absolute path (suffix it otherwise). + en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes + issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff + uploaded by qwell (license 4) Tested by: kuj, qwell + +2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp + + * apps/app_voicemail.c: Forgot the 1.4 branch for russian language + fix. (closes issue #12404) Reported by: IgorG Patches: + voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20) + + * apps/app_meetme.c: Create the directory where name recordings + will go if it does not exist. (closes issue #12311) Reported by: + rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) + + * channels/chan_sip.c: Don't add custom URI options if they don't + exist OR they are empty. (closes issue #12407) Reported by: + homesick Patches: uri_options-1.4.diff uploaded by homesick + (license 91) + +2008-04-09 20:54 +0000 [r113927] Mark Michelson + + * channels/chan_sip.c: We need to set the persistant_route [sic] + parameter for the sip_pvt during the initial INVITE, no matter if + we're building the route set from an INVITE request or response. + (closes issue #12391) Reported by: benjaminbohlmann Tested by: + benjaminbohlmann + +2008-04-09 18:57 +0000 [r113874] Tilghman Lesher + + * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does + not exist in cdr.conf, then an unload/load sequence is needed to + correct the problem. Track whether the load succeeded with a + variable, so we can fix this with a simple reload event, instead. + +2008-04-09 16:50 +0000 [r113784] Joshua Colp + + * channels/chan_iax2.c: If we receive an AUTHREQ from the remote + server and we are unable to reply (for example they have a secret + configured, but we do not) then queue a hangup frame on the + Asterisk channel. This will cause the channel to hangup and a + HANGUP to be sent via IAX2 to the remote side which is the proper + thing to do in this scenario. (closes issue #12385) Reported by: + viraptor + +2008-04-09 14:40 +0000 [r113681] Mark Michelson + + * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not + a reinvite), then we should not send a BYE. (closes issue #12392) + Reported by: fnordian Patches: chan_sip.patch uploaded by + fnordian (license 110) with small modification from me + +2008-04-09 01:34 +0000 [r113596] Terry Wilson + + * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind + happy + +2008-04-08 19:07 +0000 [r113507] Mark Michelson + + * apps/app_parkandannounce.c: Fix potential buffer overflow that + could happen if more than 100 announce files were specified when + calling ParkAndAnnounce. This overflow is not exploitable + remotely and so there is no need for a security advisory. (closes + issue #12386) Reported by: davidw + +2008-04-08 18:48 +0000 [r113402-113504] Jason Parker + + * channels/chan_skinny.c: Add a little more that is required for + previously added devices. + + * channels/chan_skinny.c: Add support for several new(ish) devices + - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver + for providing me the required information. + + * main/asterisk.c: Work around some silliness caused by + sys/capability.h - this should fix compile errors a number of + users have been experiencing. + +2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher + + * contrib/scripts/astgenkey.8: Add security note on astgenkey's + manpage. (closes issue #12373) Reported by: lmamane Patches: + 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14) + + * channels/chan_sip.c: Move check for still-bridged channels out a + little further, to avoid possible deadlocks. (Closes issue + #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt + uploaded by Corydon76 (license 14) Tested by: callguy + +2008-04-08 15:03 +0000 [r113296] Joshua Colp + + * include/asterisk/slinfactory.h, main/slinfactory.c, + main/audiohook.c: If audio suddenly gets fed into one side of a + channel after a lapse of frames flush the other factory so that + old audio does not remain in the factory causing the sync code to + not execute. (closes issue #12296) Reported by: jvandal + +2008-04-07 21:34 +0000 [r113240] Jeff Peeler + + * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This + fixes a for loop (in realtime_peer) to check all the + ast_variables the loop was intending to test rather than just the + first one. The change exposed the problem of calling memcpy on a + NULL pointer, in this case the passed in sockaddr_in struct which + is now checked. + +2008-04-07 18:00 +0000 [r113118] Jason Parker + + * channels/chan_skinny.c, configs/skinny.conf.sample: Allow + playback with noanswer (and add earlyrtp option). (closes issue + #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn + (license 30) Tested by: pj, qwell, DEA, wedhorn + +2008-04-07 17:51 +0000 [r113117] Tilghman Lesher + + * funcs/func_strings.c: Force ast_mktime() to check for DST, since + strptime(3) does not. (Closes issue #12374) + +2008-04-07 16:08 +0000 [r113065] Mark Michelson + + * main/channel.c: This fix prevents a deadlock that was experienced + in chan_local. There was deadlock prevention in place in + chan_local, but it would not work in a specific case because the + channel was recursively locked. By unlocking the channel prior to + calling the generator's generate callback in + ast_read_generator_actions(), we prevent the recursive locking, + and therefore the deadlock. (closes issue #12307) Reported by: + callguy Patches: 12307.patch uploaded by putnopvut (license 60) + Tested by: callguy + +2008-04-07 15:16 +0000 [r113012] Jeff Peeler + + * channels/chan_sip.c: (closes issue #12362) (closes issue #12372) + Reported by: vinsik Tested by: tecnoxarxa This one line change + makes an if inside a for loop (in realtime_peer) check all the + ast_variables the loop was intending to test rather than just the + first one. + +2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan + + * channels/chan_gtalk.c: Free newly allocated channel before + returning + + * channels/chan_gtalk.c: Prevent call connections when codecs don't + match. (closes issue #10604) Reported by: keepitcool Patches: + branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested + by: phsultan + +2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp + + * main/Makefile: Pass in the path to Zaptel for systems that + install Zaptel headers in a separate location. + + * main/asterisk.c: One thing at a time... let's get 1.4 building. + +2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard + + * main/asterisk.c: add a Zaptel timer check to verify the timer is + responding when Zaptel support is compiled into Asterisk and + Zaptel drivers are loaded. This will help people not waste their + valuable time debugging side effects. + +2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson + + * channels/chan_zap.c: Fix the testing of the "res" variable so + that it is more logically correct and makes the correct warning + and debug messages print. (closes issue #12361) Reported by: + one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 + (license 23) + + * main/manager.c: Fix a race condition in the manager. It is + possible that a new manager event could be appended during a + brief time when the manager is not waiting for input. If an event + comes during this period, we need to set an indicator that there + is an event pending so that the manager doesn't attempt to wait + forever for an event that already happened. (closes issue #12354) + Reported by: bamby Patches: manager_race_condition.diff uploaded + by bamby (license 430) (comments added by me) + + * apps/app_queue.c: Ensure that there is no timeout if none is + specified. (closes issue #12349) Reported by: johnlange + +2008-04-01 Russell Bryant + + * Asterisk 1.4.19 released. + +2008-03-28 Russell Bryant + + * Asterisk 1.4.19-rc4 released. + +2008-03-28 16:19 +0000 [r111658] Jason Parker + + * formats/format_wav_gsm.c: The file size of WAV49 does not need to + be an even number. (closes issue #12128) Reported by: mdu113 + Patches: 12128-noevenlength.diff uploaded by qwell (license 4) + Tested by: qwell, mdu113 + +2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher + + * doc/valgrind.txt: Update debugging text, since Valgrind + eliminated the --log-file-exactly option. (Closes issue #12320) + + * main/acl.c: For FreeBSD, at least, the ifa_addr element could be + NULL. (closes issue #12300) Reported by: festr Patches: + acl.c.patch uploaded by festr (license 443) + +2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy + + * apps/app_playback.c, main/pbx.c: These small documentation + updates made in response to a query in asterisk-users, where a + user was using Playback, but needed the features of Background, + and had no idea that Background existed, or that it might provide + the features he needed. I thought the best way to avert these + kinds of queries was to provide "See Also" references in all + three of "Background", "Playback", "WaitExten". Perhaps a project + to do this with all related apps is in order. + + * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue + #12302) Reported by: pj Tested by: murf These changes will set a + channel variable ~~EXTEN~~ just before generating code for a + switch, with the value of ${EXTEN}. The exten is marked as having + a switch, and ever after that, till the end of the exten, we + substitute any ${EXTEN} with ${~~EXTEN~~} instead in application + arguments; (and the ${EXTEN: also). The reason for this, is that + because switches are coded using separate extensions to provide + pattern matching, and jumping to/from these switch extensions + messes up the ${EXTEN} value, which blows the minds of users. + +2008-03-27 00:25 +0000 [r111245-111280] Jason Parker + + * main/frame.c: Put this flag back so we don't change the API. + + * main/frame.c: Remove excessive smoother optimization that was + causing audio glitches (small "pops") after (about 200ms later) + an "incorrectly" sized frame was received. While it would be very + nice to keep this as optimized as possible, it makes no sense for + the smoother to be dropping random bits of audio like this. Isn't + that the whole point of a smoother? Closes issue #12093. + +2008-03-26 19:55 +0000 [r111129] Joshua Colp + + * contrib/scripts/autosupport: Update autosupport script. (closes + issue #12310) Reported by: angler Patches: autosupport.diff + uploaded by angler (license 106) + +2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming + + * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar + 2008) | 2 lines update UPGRADE notes to document usage of the + script ........ + +2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson + + * apps/app_voicemail.c: This code change is made just for + clarification. It does exactly the same thing as before. It just + doesn't look as wrong. + + * apps/app_voicemail.c: Add a lock to the vm_state structure and + use the lock around mail_open calls to prevent concurrent access + of the same mailstream. This, along with trunk's ability to + configure TCP timeouts for IMAP storage will help to prevent + crashes and hangs when using voicemail with IMAP storage. (closes + issue #10487) Reported by: ewilhelmsen + +2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming + + * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): + Merged revisions 111019 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar + 2008) | 2 lines add a script to make getting the iLBC source code + simple for end users ........ + +2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp + + * channels/chan_sip.c: If we are requested to authenticate a + reinvite make sure that it contains T38 SDP if need be. (closes + issue #11995) Reported by: fall + + * channels/chan_iax2.c: Make sure that full video frames are sent + whenever the 15 bit timestamp rolls over. (closes issue #11923) + Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by + mihai (license 94) + +2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming + + * UPGRADE.txt: add note that the user will need to enable + codec_ilbc to get it to build + + * codecs/ilbc/StateConstructW.h (removed), + codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h + (removed), codecs/ilbc/getCBvec.c (removed), + codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c + (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c + (removed), codecs/ilbc/getCBvec.h (removed), + codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h + (removed), codecs/ilbc/FrameClassify.c (removed), + codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), + codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h + (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c + (removed), codecs/ilbc/anaFilter.c (removed), + codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c + (removed), codecs/ilbc/doCPLC.h (removed), + codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, + codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c + (removed), codecs/ilbc/createCB.h (removed), CHANGES, + codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h + (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed), + codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), + codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c + (removed), codecs/ilbc/iCBSearch.h (removed), + codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), + codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c + (removed), codecs/ilbc/hpOutput.h (removed), + codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, + codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h + (removed), codecs/ilbc/iCBConstruct.c (removed), + codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h + (removed), codecs/ilbc/syntFilter.h (removed), + codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c + (removed): Merged revisions 110869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar + 2008) | 2 lines due to licensing restrictions, we cannot + distribute the source code for iLBC encoding and decoding... so + remove it, and add instructions on how the user can obtain it + themselves ........ + +2008-03-25 22:51 +0000 [r110779] Jason Parker + + * cdr/cdr_custom.c: Make file access in cdr_custom similar to + cdr_csv. Fixes issue #12268. Patch borrowed from r82344 + +2008-03-25 20:03 +0000 [r110727] Jeff Peeler + + * channels/chan_sip.c: This one line change makes an if inside a + for loop (in realtime_peer) check all the ast_variables the loop + was intending to test rather than just the first one. + +2008-03-25 15:40 +0000 [r110635] Mark Michelson + + * channels/chan_sip.c: When reverting a commit, I accidentally left + in this bit which was an experiment to see what would happen. It + passed the compile test, and I didn't notice I had left this + change in too. So this is a revert of a revert...sort of. + +2008-03-25 14:37 +0000 [r110628] Joshua Colp + + * include/asterisk/options.h, main/asterisk.c, Makefile, + main/app.c: Add an option (transmit_silence) which transmits + silence during both Record() and DTMF generation. The reason this + is an option is that in order to transmit silence we have to + setup a translation path. This may not be needed/wanted in all + cases. (closes issue #10058) Reported by: tracinet + +2008-03-24 19:17 +0000 [r110618] Mark Michelson + + * channels/chan_sip.c: This is a revert for revision 108288. The + reason is that that revision was not for an actual bug fix per + se, and so it really should not have been in 1.4 in the first + place. Plus, people who compile with DO_CRASH are more likely to + encounter a crash due to this change. While I think the usage of + DO_CRASH in ast_sched_del is a bit absurd, this sort of change is + beyond the scope of 1.4 and should be done instead in a developer + branch based on trunk so that all scheduler functions are fixed + at once. I also am reverting the change to trunk and 1.6 since + they also suffer from the DO_CRASH potential. (closes issue + #12272) Reported by: qq12345 + +2008-03-24 17:34 +0000 [r110614] Russell Bryant + + * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message. + +2008-03-21 14:32 +0000 [r110474] Jason Parker + + * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on + mips platforms either. (closes issue #12270) Reported by: + zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt + (license 33) + +2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant + + * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms + to 50 ms in the autoservice thread. This really should not make a + difference except in very rare cases. That case would be that all + of the channels in autoservice are not generating any frames. In + that case, this change reduces the potential amount of time that + a thread waits in ast_autoservice_stop() for the autoservice + thread to wrap back around to the beginning of its loop. (closes + issue #12266, reported by dimas) + + * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions + 110335 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) + | 6 lines Fix some very broken code that was introduced in 1.2.26 + as a part of the security fix. The dnsmgr is not appropriate + here. The dnsmgr takes a pointer to an address structure that a + background thread continuously updates. However, in these cases, + a stack variable was passed. That means that the dnsmgr thread + would be continuously writing to bogus memory. ........ + + * apps/app_meetme.c: Fix a bug where when calls on the trunk side + hang up while on hold, the state is not properly reflected. + (closes issue #11990, reported by anakaoka, patched by me) + +2008-03-19 20:33 +0000 [r110083] Mark Michelson + + * apps/app_chanspy.c: Add a missing unlock in the case that memory + allocation fails in app_chanspy. Thanks to Russell for confirming + that this was an issue. + +2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp + + * res/res_musiconhold.c: Add sanity checking for position resuming. + We *have* to make sure that the position does not exceed the + total number of files present, and we have to make sure that the + position's filename is the same as previous. These values can + change if a music class is reloaded and give unpredictable + behavior. (closes issue #11663) Reported by: junky + + * main/rtp.c: Make sure that the mark bit does not incorrectly + cause video frame timestamps to be calculated as if they are + audio frames. (closes issue #11429) Reported by: sperreault + Patches: 11429-frametype.diff uploaded by qwell (license 4) + +2008-03-19 17:12 +0000 [r109973] Jason Parker + + * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml + (added): People report bugs about Asterisk crashing with DO_CRASH + enabled was getting a little silly... Now we only show certain + cflags when you run configure with --enable-dev-mode + (corresponding menuselect change to follow) + +2008-03-19 15:41 +0000 [r109908] Steve Murphy + + * main/config.c: (closes issue #11442) Reported by: tzafrir + Patches: 11442.patch uploaded by murf (license 17) Tested by: + murf I didn't give tzafrir very much time to test this, but if he + does still have remaining issues, he is welcome to re-open this + bug, and we'll do what is called for. I reproduced the problem, + and tested the fix, so I hope I am not jumping by just going + ahead and committing the fix. The problem was with what file_save + does with templates; firstly, it tended to print out multiple + options: [my_category](!)(templateref) instead of + [my_category](!,templateref) which is fixed by this patch. + Nextly, the code to suppress output of duplicate declarations + that would occur because the reader copies inherited declarations + down the hierarchy, was not working. Thus: [master-template](!) + mastervar = bar [template](!,master-template) tvar = value + [cat](template) catvar = val would be rewritten as: ;! ;! + Automatically generated configuration file ;! Filename: + experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: + Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! + [master-template](!) mastervar = bar + [template](!,master-template) mastervar = bar tvar = value + [cat](template) mastervar = bar tvar = value catvar = val This + has been fixed. Since the config reader 'explodes' inherited vars + into the category, users may, in certain circumstances, see + output different from what they originally entered, but it should + be both correct and equivalent. + +2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant + + * main/utils.c: Tweak spacing in a recent change because I'm very + picky. + + * apps/app_chanspy.c: Fix one place where the chanspy datastore + isn't removed from a channel. (issue #12243, reported by atis, + patch by me) + +2008-03-18 20:52 +0000 [r109713] Mark Michelson + + * apps/app_queue.c: This patch makes it so that all queue member + status changes are handled through device state code. This + removes several problems people were seeing where their queue + members would get into an "unknown" state. Huge props go to atis + on this one since he was the one who found the code section that + was causing the problem and proposed the solution. I just wrote + what he suggested :) (closes issue #12127) Reported by: atis + Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested + by: atis, jvandal + +2008-03-18 19:23 +0000 [r109648] Jason Parker + + * codecs/log2comp.h: Allow codecs that use log2comp (g726) to + compile correctly on x86 with gcc4 optimizations. (closes issue + #12253) Reported by: fossil Patches: log2comp.patch uploaded by + fossil (license 140) + +2008-03-18 17:58 +0000 [r109575] Mark Michelson + + * channels/chan_agent.c: Make sure an agent doesn't try to send + dtmf to a NULL channel closes issue #12242 Reported by Yourname + +2008-03-18 Russell Bryant + + * Asterisk 1.4.19-rc3 released. + +2008-03-18 16:25 +0000 [r109482] Terry Wilson + + * include/asterisk/astobj.h: Fix character string being treated ad + format string + +2008-03-18 15:10 +0000 [r109393] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) | + 3 lines Do not return with a successful authentication if the + From header ends up empty. (AST-2008-003) ........ + +2008-03-18 14:58 +0000 [r109386] Joshua Colp + + * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the + number of payloads accepted, and also make sure a given payload + does not exceed our maximum value. (AST-2008-002) + +2008-03-18 06:37 +0000 [r109309] Steve Murphy + + * pbx/ael/ael-test/ael-ntest23 (added), + pbx/ael/ael-test/ael-ntest23/t1/a.ael (added), + pbx/ael/ael-test/ael-ntest23/t1/b.ael (added), + pbx/ael/ael-test/ael-ntest23/t1/c.ael (added), + pbx/ael/ael-test/ael-ntest23/t2/d.ael (added), + pbx/ael/ael-test/ael-ntest23/t2/e.ael (added), + pbx/ael/ael-test/ael-ntest23/t2/f.ael (added), + pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c, + pbx/ael/ael-test/ael-ntest23/t3/g.ael (added), + pbx/ael/ael-test/ael-ntest23/t3/h.ael (added), + pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex, + pbx/ael/ael-test/ael-ntest23/t3/j.ael (added), + pbx/ael/ael-test/ael-ntest23/qq.ael (added), + pbx/ael/ael-test/ael-ntest23/t1 (added), + pbx/ael/ael-test/ael-ntest23/t2 (added), + pbx/ael/ael-test/ael-ntest23/t3 (added), + pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes + issue #11903) Reported by: atis Many thanks to atis for spotting + this problem and reporting it. The fix was to straighten out how + items are placed on and removed from the file stack. Regressions + as well as the provided test case helped to straighten out all + code paths. valgrind was used to make sure all memory allocated + was freed. Sorry for not solving this earlier. I got distracted. + Added the ntest23 regression test, which is mainly a copy of + ntest22, but with a few juicy errors thrown in, to replicate the + kind of error that atis spotted. + +2008-03-17 22:05 +0000 [r109226] Mark Michelson + + * main/utils.c: Fix a logic flaw in the code that stores lock info + which is displayed via the "core show locks" command. The idea + behind this section of code was to remove the previous lock from + the list if it was a trylock that had failed. Unfortunately, + instead of checking the status of the previous lock, we were + referencing the index immediately following the previous lock in + the lock_info->locks array. The result of this problem, under the + right circumstances, was that the lock which we currently in the + process of attempting to acquire could "overwrite" the previous + lock which was acquired. While this does not in any way affect + typical operation, it *could* lead to misleading "core show + locks" output. + +2008-03-17 17:55 +0000 [r109171] Michiel van Baak + + * channels/chan_skinny.c: Update the directory of placed calls on + skinny phones when dialing a channel that does not provide + progress (analog ZAP lines) The phone does handle the double + update on calls to channels that do provide progress and wont + insert duplicate items (closes issue #12239) Reported by: DEA + Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) + +2008-03-17 16:24 +0000 [r109107] Joshua Colp + + * channels/chan_sip.c: 200 OKs in response to a reinvite need to be + sent reliably. If the remote side does not receive one the dialog + will be torn down. (closes issue #12208) Reported by: atrash + +2008-03-17 15:15 +0000 [r109057] Jason Parker + + * main/file.c: Backport revision 106439 from trunk. I didn't + realize this was broken in 1.4 as well. Closes issue #12222. + +2008-03-17 14:18 +0000 [r109012] Mark Michelson + + * apps/app_chanspy.c: Make sure that we release the lock on the + spyee channel if the spyee or spy has hung up (closes issue + #12232) Reported by: atis + +2008-03-16 21:47 +0000 [r108961] Michiel van Baak + + * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE + (closes issue #12228) Reported by: andrew Patches: SRC.patch + uploaded by andrew (license 240) + +2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant + + * channels/chan_oss.c: Fix a channel name issue. chan_oss registers + the "Console" channel type, but it created channels with an "OSS" + prefix. (closes issue #12194, reported by davidw, patched by me) + + * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to + start networking before asterisk, as well. (closes issue #12200, + reported by and change suggested by reinerotto) + +2008-03-14 16:44 +0000 [r108737] Mark Michelson + + * channels/chan_sip.c: Fix a race condition in the SIP packet + scheduler which could cause a crash. chan_sip uses the scheduler + API in order to schedule retransmission of reliable packets (such + as INVITES). If a retransmission of a packet is occurring, then + the packet is removed from the scheduler and retrans_pkt is + called. Meanwhile, if a response is received from the packet as + previously transmitted, then when we ACK the response, we will + remove the packet from the scheduler and free the packet. The + problem is that both the ACK function and retrans_pkt attempt to + acquire the same lock at the beginning of the function call. This + means that if the ACK function acquires the lock first, then it + will free the packet which retrans_pkt is about to read from and + write to. The result is a crash. The solution: 1. If the ACK + function fails to remove the packet from the scheduler and the + retransmit id of the packet is not -1 (meaning that we have not + reached the maximum number of retransmissions) then release the + lock and yield so that retrans_pkt may acquire the lock and + operate. 2. Make absolutely certain that the ACK function does + not recursively lock the lock in question. If it does, then + releasing the lock will do no good, since retrans_pkt will still + be unable to acquire the lock. (closes issue #12098) Reported by: + wegbert (closes issue #12089) Reported by: PTorres Patches: + 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested + by: jvandal + +2008-03-14 14:29 +0000 [r108682] Jason Parker + + * res/res_musiconhold.c: Fix a potential segfault if chan (or + chan->music_state) is NULL. Closes issue #12210, credit to + edantie for pointing this out. + +2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant + + * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h: + Fix another issue that was causing crashes in chanspy. This + introduces a new datastore callback, called chan_fixup(). The + concept is exactly like the fixup callback that is used in the + channel technology interface. This callback gets called when the + owning channel changes due to a masquerade. Before this was + introduced, if a masquerade happened on a channel being spyed on, + the channel pointer in the datastore became invalid. (closes + issue #12187) (reported by, and lots of testing from atis) (props + to file for the help with ideas) + + * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom + phones to blink when an extension that has been subscribed to + goes on hold. Otherwise, they just stay on like it does when an + extension is in use. (closes issue #11263) Reported by: russell + Patches: notify_hold.rev1.txt uploaded by russell (license 2) + Tested by: russell + + * apps/app_followme.c: Fix a couple uses of sprintf. The second one + could actually cause an overflow of a stack buffer. It's not a + security issue though, it only depends on your configuration. + +2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson + + * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del + for autocongestion in chan_sip. The scheduler callback will + always return 0. This means that this id is never rescheduled, so + it makes no sense to loop trying to delete the id from the + scheduler queue. If we fail to remove the item from the queue + once, it will fail every single time. (Yes I realize that in this + case, the macro would exit early because the id is set to -1 in + the callback, but it still makes no sense to use that macro in + favor of calling ast_sched_del once and being done with it) This + is the first of potentially several such fixes. + + * include/asterisk/sched.h: Added a large comment before the + AST_SCHED_DEL macro to explain its purpose as well as when it is + appropriate and when it is not appropriate to use it. I also + removed the part of the debug message that mentions that this is + probably a bug because there are some perfectly legitimate places + where ast_sched_del may fail to delete an entry (e.g. when the + scheduler callback manually reschedules with a new id instead of + returning non-zero to tell the scheduler to reschedule with the + same idea). I also raised the debug level of the debug message in + AST_SCHED_DEL since it seems like it could come up quite + frequently since the macro is probably being used in several + places where it shouldn't be. Also removed the redundant line, + file, and function information since that is provided by ast_log. + +2008-03-12 19:57 +0000 [r108135] Russell Bryant + + * apps/app_chanspy.c, main/channel.c: (closes issue #12187, + reported by atis, fixed by me after some brainstorming on the + issue with mmichelson) - Update copyright info on app_chanspy. - + Fix a race condition that caused app_chanspy to crash. The issue + was that the chanspy datastore magic that was used to ensure that + spyee channels did not disappear out from under the code did not + completely solve the problem. It was actually possible for + chanspy to acquire a channel reference out of its datastore to a + channel that was in the middle of being destroyed. That was + because datastore destruction in ast_channel_free() was done near + the end. So, this left the code in app_chanspy accessing a + channel that was partially, or completely invalid because it was + in the process of being free'd by another thread. The following + sort of shows the code path where the race occurred: + ============================================================================= + Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) + --------------------------------------||------------------------------------- + ast_channel_free() || - remove channel from channel list || - + lock/unlock the channel to ensure || that no references retrieved + from || the channel list exist. || + --------------------------------------||------------------------------------- + || channel_spy() - destroy some channel data || - Lock chanspy + datastore || - Retrieve reference to channel || - lock channel || + - Unlock chanspy datastore + --------------------------------------||------------------------------------- + - destroy channel datastores || - call chanspy datastore d'tor || + which NULL's out the ds' || - Operate on the channel ... + reference to the channel || || - free the channel || || || - + unlock the channel + --------------------------------------||------------------------------------- + ============================================================================= + +2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming + + * channels/chan_sip.c: if we receive an INVITE with a + Content-Length that is not a valid number, or is zero, then don't + process the rest of the message body looking for an SDP closes + issue #11475 Reported by: andrebarbosa + +2008-03-12 18:26 +0000 [r108083] Joshua Colp + + * apps/app_mixmonitor.c, include/asterisk/audiohook.h, + main/audiohook.c: Add a trigger mode that triggers on both read + and write. The actual function that returns the combined audio + frame though will wait until both sides have fed in audio, or + until one side stops (such as the case when you call Wait). + (closes issue #11945) Reported by: xheliox + +2008-03-12 16:59 +0000 [r108031] Russell Bryant + + * main/channel.c: Destroy the channel lock after the channel + datastores. (inspired by issue #12187) + +2008-03-12 01:52 +0000 [r107877] Tilghman Lesher + + * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql: + Document all of the possible realtime fields + +2008-03-11 23:37 +0000 [r107714-107826] Jason Parker + + * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql + ODBC voicemail. (closes issue #12186) Reported by: jsmith + Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license + 15) + + * channels/chan_gtalk.c: Copy voicemail dependency logic for + res_adsi to chan_gtalk (for jabber). (closes issue #12014) + Reported by: junky + +2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming + + * Makefile.rules, channels/Makefile: get chan_vpb to build properly + in dev mode + +2008-03-11 20:47 +0000 [r107712] Jason Parker + + * apps/app_voicemail.c: Add a newline on a log + +2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp + + * res/res_features.c: Make sure the visible indication is on the + right channel so when the masquerade happens the proper + indication is enacted. (closes issue #11707) Reported by: iam + + * apps/app_meetme.c: Add an additional check for setting conference + parameter when using the marked user options. It was possible for + it to return to a no listen/no talk state if a masquerade + happened. (closes issue #12136) Reported by: aragon + + * apps/app_exec.c: Fix a minor spelling error. (closes issue + #12183) Reported by: darrylc + +2008-03-11 Russell Bryant + + * Asterisk 1.4.19-rc2 released. + +2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming + + * apps/app_rpt.c: backport a fix from trunk + + * channels/misdn/isdn_lib.c, codecs/Makefile, + channels/chan_misdn.c: fix various other problems found by gcc + 4.3 + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + apps/app_sms.c: stop checking for mktime() in the configure + script... we don't use it, and the test is buggy under gcc 4.3 + + * configure, main/Makefile, configure.ac, makeopts.in: check for + compiler support for -fno-strict-overflow before using it (tested + with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179) + Reported by: Netview + + * configure, configure.ac: fix small bug in IMAP toolkit testing + + * main/udptl.c, utils/Makefile, main/Makefile, + main/editline/readline.c, pbx/Makefile: fix up various compiler + warnings found with gcc-4.3: - the output of flex includes a + static function called 'input' that is not used, so for the + moment we'll stop having the compiler tell us about unused + variables in the flex source files (a better fix would be to + improve our flex post-processing to remove the unused function) - + main/stdtime/localtime.c makes assumptions about signed integer + overflow, and gcc-4.3's improved optimizer tries to take + advantage of handling potential overflow conditions at compile + time; for now, suppress these optimizations until we can fiure + out if the code needs improvement - main/udptl.c has some + references to uninitialized variables; in one case there was no + bug, but in the other it was certainly possibly for unexpected + behavior to occur - main/editline/readline.c had an unused + variable + +2008-03-11 00:59 +0000 [r107290] Terry Wilson + + * channels/chan_sip.c: If we fail to alloc a channel, we should + re-lock the pvt structure before returning. + +2008-03-10 21:32 +0000 [r107230] Tilghman Lesher + + * main/pbx.c: Use non-global storage for eswitch + +2008-03-10 20:27 +0000 [r107173] Jason Parker + + * channels/chan_zap.c: Make sure to reenable echo can after a + "failed" (canceled, etc) three-way call. (closes issue #11335) + Reported by: rebuild + +2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant + + * main/pbx.c: Fix another bug specifically related to asynchronous + call origination. Once the PBX is started on the channel using + ast_pbx_start(), then the ownership of the channel has been + passed on to another thread. We can no longer access it in this + code. If the channel gets hung up very quickly, it is possible + that we could access a channel that has been free'd. (inspired by + BE-386) + + * main/pbx.c: Fix some bugs related to originating calls. If the + code failed to start a PBX on the channel (such as if you set a + call limit based on the system's load average), then there were + cases where a channel that has already been free'd using + ast_hangup() got accessed. This caused weird memory corruption + and crashes to occur. (fixes issue BE-386) (much debugging credit + goes to twilson, final patch written by me) + + * main/channel.c: Resolve a compiler warning. + + * main/channel.c: Fix a race condition where the generator can go + away (closes issue #12175, reported by edantie, patched by me) + +2008-03-10 14:33 +0000 [r107016] Joshua Colp + + * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where + unanswered CDRs are dropped to the CDR core, not everything uses + app_dial. (closes issue #11516) Reported by: ys Patches: + branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: + anest, jcapp, dartvader + +2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming + + * channels/chan_zap.c: don't generate D-Channel "up" and "down" + messages unless the channel state is actually changing; also, + generate the "up" message when an implicit "up" occurs due to + reception of a normal event when we thought the channel was + "down" + +2008-03-07 22:51 +0000 [r106895] Russell Bryant + + * apps/app_meetme.c: Only start the SLA thread if SLA has actually + been configured. + +2008-03-07 22:14 +0000 [r106842] Jason Parker + + * main/editline/Makefile.in: Fix hardcoded grep in editline, were + GNU grep is required. (closes issue #12124) Reported by: dmartin + +2008-03-07 19:32 +0000 [r106788] Joshua Colp + + * main/channel.c: Ignore source update control frame. (closes issue + #12168) Reported by: plack + +2008-03-07 17:16 +0000 [r106704] Russell Bryant + + * include/asterisk/sched.h: Change a warning message to a debug + message. This is happening quite frequently, and it is not worth + spamming users with these messages unless we are pretty confident + that it should never happen. As it stands today, it _will_ and + _does_ happen and until that gets cleaned up a reasonable amount + on the development side, let's not spam the logs of everyone + else. (closes issue #12154) + +2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher + + * apps/app_voicemail.c: Warn the user when a temporary greeting + exists (Closes issue #11409) + + * main/rtp.c: Properly initialize rtp->schedid (Closes issue + #12154) + + * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, + apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, + funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, + main/manager.c: Safely use the strncat() function. (closes issue + #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt + uploaded by Corydon76 (license 14) + +2008-03-06 22:10 +0000 [r106437] Mark Michelson + + * main/pbx.c: Quell an annoying message that is likely to print + every single time that ast_pbx_outgoing_app is called. The reason + is that __ast_request_and_dial allocates the cdr for the channel, + so it should be expected that the channel will have a cdr on it. + Thanks to joetester on IRC for pointing this out + +2008-03-06 04:40 +0000 [r106328] Tilghman Lesher + + * sounds/Makefile: Upgrade to the next release of sounds + +2008-03-05 22:37 +0000 [r106237] Russell Bryant + + * channels/chan_iax2.c: Fix a potential deadlock and a few + different potential crashes. (closes issue #12145, reported by + thiagarcia, patched by me) + +2008-03-05 22:32 +0000 [r106235] Joshua Colp + + * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c, + apps/app_dial.c, main/channel.c, channels/chan_phone.c, + main/dial.c, channels/chan_zap.c, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, main/file.c, + channels/chan_alsa.c, apps/app_followme.c, + include/asterisk/frame.h: Add a control frame to indicate the + source of media has changed. Depending on the underlying + technology it may need to change some things. (closes issue + #12148) Reported by: jcomellas + +2008-03-05 21:12 +0000 [r106178] Michiel van Baak + + * doc/realtime.txt: document var_metric so no bugreports will come + in when it's actually a configuration issue. (issue #12151) + Reported and patched by: caio1982 1.4 patch by me + +2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming + + * channels/chan_zap.c: when a PRI call must be moved to a different + B channel at the request of the other endpoint, ensure that any + DSP active on the original channel is moved to the new one + (closes issue #11917) Reported by: mavetju Tested by: mavetju + +2008-03-05 15:17 +0000 [r106015] Tilghman Lesher + + * channels/chan_sip.c, include/asterisk/sched.h: Correctly + initialize retransid in SIP, and ensure that the warning when + failing to delete a schedule entry can actually hit the log. + (closes issue #12140) Reported by: slavon Patches: sch2.patch + uploaded by slavon (license 288) (Patch slightly modified by me) + +2008-03-05 01:52 +0000 [r105932] Russell Bryant + + * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug + that I just noticed in the RTP code. The calculation for setting + the len field in an ast_frame of audio was wrong when G.722 is in + use. The len field represents the number of ms of audio that the + frame contains. It would have set the value to be twice what it + should be. + +2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp + + * main/rtp.c: In addition to setting the marker bit let's change + our ssrc so they know for sure it is a different source. + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a + new source of audio comes in (such as music on hold) make sure + the marker bit gets set. (closes issue #10355) Reported by: + wdecarne Patches: 10355.diff uploaded by file (license 11) + (closes issue #11491) Reported by: kanderson + +2008-03-04 Russell Bryant + + * Asterisk 1.4.19-rc1 released. + +2008-03-04 04:31 +0000 [r105591] Russell Bryant + + * main/pbx.c: Backport a minor bug fix from trunk that I found + while doing random code cleanup. Properly break out of the loop + when a context isn't found when verify that includes are valid. + +2008-03-03 18:06 +0000 [r105572] Jason Parker + + * res/snmp/agent.c: Fix type for astNumChannels. (closes issue + #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg + (license 192) + +2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant + + * channels/chan_local.c: In the case of an ast_channel allocation + failure, take the local_pvt out of the pvt list before destroying + it. + + * channels/chan_local.c: Fix a potential memory leak of the + local_pvt struct when ast_channel allocation fails. Also, in + passing, centralize the code necessary to destroy a local_pvt. + + * main/autoservice.c: Update the copyright information for + autoservice. Most of the code in this file now is stuff that I + have written recently ... + + * main/asterisk.c, main/channel.c, include/asterisk.h, + main/autoservice.c: Merge in some changes from + team/russell/autoservice-nochans-1.4 These changes fix up some + dubious code that I came across while auditing what happens in + the autoservice thread when there are no channels currently in + autoservice. 1) Change it so that autoservice thread doesn't keep + looping around calling ast_waitfor_n() on 0 channels twice a + second. Instead, use a thread condition so that the thread + properly goes to sleep and does not wake up until a channel is + put into autoservice. This actually fixes an interesting bug, as + well. If the autoservice thread is already running (almost always + is the case), then when the thread goes from having 0 channels to + have 1 channel to autoservice, that channel would have to wait + for up to 1/2 of a second to have the first frame read from it. + 2) Fix up the code in ast_waitfor_nandfds() for when it gets + called with no channels and no fds to poll() on, such as was the + case with the previous code for the autoservice thread. In this + case, the code would call alloca(0), and pass the result as the + first argument to poll(). In this case, the 2nd argument to + poll() specified that there were no fds, so this invalid pointer + shouldn't actually get dereferenced, but, this code makes it + explicit and ensures the pointers are NULL unless we have valid + data to put there. (related to issue #12116) + +2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp + + * main/channel.c: It is possible for no audio to pass between the + current digit and next digit so expand logic that clears + emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: + edgreenberg Patches: v1-11911.patch uploaded by dimas (license + 88) Tested by: tbsky + + * channels/chan_sip.c: Add a comment to describe some logic. + (closes issue #12120) Reported by: flefoll Patches: + chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license + 244) + +2008-02-29 23:34 +0000 [r105409] Russell Bryant + + * main/autoservice.c: Fix a major bug in autoservice. There was a + race condition in the handling of the list of channels in + autoservice. The problem was that it was possible for a channel + to get removed from autoservice and destroyed, while the + autoservice thread was still messing with the channel. This led + to memory corruption, and caused crashes. This explains multiple + backtraces I have seen that have references to autoservice, but + do to the nature of the issue (memory corruption), could cause + crashes in a number of areas. (fixes the crash in BE-386) (closes + issue #11694) (closes issue #11940) The following issues could be + related. If you are the reporter of one of these, please update + to include this fix and try again. (potentially fixes issue + #11189) (potentially fixes issue #12107) (potentially fixes issue + #11573) (potentially fixes issue #12008) (potentially fixes issue + #11189) (potentially fixes issue #11993) (potentially fixes issue + #11791) + +2008-02-29 14:47 +0000 [r105326] Philippe Sultan + + * res/res_jabber.c: Fix a potential memory leak + +2008-02-29 14:34 +0000 [r105296] Tilghman Lesher + + * apps/app_voicemail.c: If the message file does not exist, just + return harmlessly, instead of crashing. (Closes issue #12108) + +2008-02-29 13:48 +0000 [r105261] Joshua Colp + + * apps/app_voicemail.c: Bump up the size of the uniqueid variable. + (closes issue #12107) Reported by: asgaroth + +2008-02-29 13:05 +0000 [r105209] Philippe Sultan + + * res/res_jabber.c: Automatically create new buddy upon reception + of a presence stanza of type subscribed. (closes issue #12066) + Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by + phsultan (license 73) trunk-12066-1.diff uploaded by phsultan + (license 73) Tested by: ffadaie, phsultan + +2008-02-28 22:23 +0000 [r105116] Russell Bryant + + * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock + tracking code that was discovered by mmichelson. The issue is + that if the lock history array was full, then the functions to + mark a lock as acquired or not would adjust the stats for + whatever lock is at the end of the array, which may not be + itself. So, do a sanity check to make sure that we're updating + lock info for the proper lock. (This explains the bizarre stats + on lock #63 in BE-396, thanks Mark!) + +2008-02-28 21:56 +0000 [r105113] Tilghman Lesher + + * contrib/init.d/rc.debian.asterisk: Update init script for LSB + compat (closes issue #9843) Reported by: ibc Patches: + rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by: + paravoid + +2008-02-28 20:11 +0000 [r105059] Mark Michelson + + * apps/app_queue.c: When using autofill, members who are in use + should be counted towards the number of available members to call + if ringinuse is set to yes. Thanks to jmls who brought this issue + up on IRC + +2008-02-28 19:20 +0000 [r104920-105005] Jason Parker + + * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into + applications, if we get NULL. This protects against possible + segfaults in applications that may try to use data before + checking length (ast_strdupa'ing it, for example) (closes issue + #12100) Reported by: foxfire Patches: 12100-nullappargs.diff + uploaded by qwell (license 4) + + * channels/chan_skinny.c: According to a video at www.cisco.com, + the 7921G supports 6 line appearances. + +2008-02-28 00:05 +0000 [r104868] Tilghman Lesher + + * main/Makefile, build_tools/strip_nonapi: Compatibility fix for + PPC64 (closes issue #12081) Reported by: jcollie Patches: + asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) + Tested by: jcollie, Corydon76 + +2008-02-27 21:49 +0000 [r104841] Mark Michelson + + * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a + lockable list. This is because in some cases, the ast_dial may + exist in multiple threads due to asynchronous execution of its + application, and I found some cases where race conditions could + exist. 2. Check in ast_dial_join to be sure that the channel + still exists before attempting to lock it, since it could have + gotten hung up but the is_running_app flag on the + ast_dial_channel may not have been cleared yet. (closes issue + #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by + putnopvut (license 60) Tested by: jvandal + +2008-02-27 20:56 +0000 [r104787] Joshua Colp + + * apps/app_chanspy.c: Don't loop around infinitely trying to spy on + our own channel, and don't forget to free/detach the datastore + upon hangup of the spy. + +2008-02-27 20:36 +0000 [r104783] Mark Michelson + + * main/file.c: Bump a couple of more buffers up by 2 so that + annoying warnings aren't generated like crazy on every + fileexists_core call. + +2008-02-27 18:15 +0000 [r104704] Tilghman Lesher + + * main/manager.c: Ensure the session ID can't be 0. + +2008-02-27 17:41 +0000 [r104665] Joshua Colp + + * main/file.c: Bump up the buffer by 2. + +2008-02-27 17:33 +0000 [r104625] Russell Bryant + + * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get + stuck in an infinite loop without being able to detect that the + calling channel hung up. (closes issue #12076, reported by junky, + patched by me) + +2008-02-27 17:26 +0000 [r104598] Jason Parker + + * res/res_features.c: Inherit language from the transfering channel + on a blind transfer. (closes issue #11682) Reported by: caio1982 + Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 + (license 22) Tested by: caio1982, victoryure + +2008-02-27 17:07 +0000 [r104596] Joshua Colp + + * main/loader.c: Use the lock (which already existed, it just + wasn't used) on the updaters list to protect the contents instead + of the overall module list lock. (closes issue #12080) Reported + by: ChaseVenters + +2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming + + * main/file.c: fallback to standard English prompts properly when + using new prompt directory layout (closes issue #11831) Reported + by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license + 20) (modified by me to improve code and conform rest of function + to coding guidelines) + +2008-02-27 16:45 +0000 [r104591] Russell Bryant + + * channels/chan_zap.c: When we receive a known alarm, make sure + that the unknown alarm flag is not still set to make sure that + when we come back out of alarm, it gets reported in the log and + manager interface (after discussion with tzafrir on the -dev + list) + +2008-02-27 15:52 +0000 [r104536] Joshua Colp + + * res/res_smdi.c: Only stop the MWI monitor thread if it was + actually started. (closes issue #12086) Reported by: francesco_r + +2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant + + * apps/app_chanspy.c: Avoid some recursion in the cleanup code for + the chanspy datastore (closes issue #12076, reported by junky, + patched by me) + + * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as + an alarm. However, Asterisk 1.4 does not know what to do with + these alarms. Only Asterisk 1.6 cares about it. So, if we get an + unknown alarm in chan_zap, don't generate confusing log messages + about it. + +2008-02-26 18:26 +0000 [r104132-104141] Jason Parker + + * Makefile: Add badshell to .PHONY target (thanks Kevin) + + * Makefile: Since all shells aren't as awesome as bash, we have to + fail if somebody tries to use a literal "~" in DESTDIR. + + * sounds/Makefile: Revert previous abspath change. ...abspath is + new in GNU make 3.81. I feel so...defeated. Must find new fix! + + * sounds/Makefile: Fix a very bizarre issue we were seeing with our + buildbot when using a DESTDIR that wasn't an absolute path (such + as DESTDIR=~/asterisk-1.4). Apparently what was happening, was + that some of the targets were being expanded to the full path, so + $@ ended up being /root/asterisk-1.4/[...]/ rather than + ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" + in GNU make. (*cough* + http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) + +2008-02-26 00:25 +0000 [r104119] Russell Bryant + + * include/asterisk/smdi.h, apps/app_voicemail.c, + channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample: + Merge changes from team/russell/smdi-1.4 This commit brings in a + significant set of changes to the SMDI support in Asterisk. There + were a number of bugs in the current implementation, most notably + being that it was very likely on busy systems to pop off the + wrong message from the SMDI message queue. So, this set of + changes fixes the issues discovered as well as introducing some + new ways to use the SMDI support which are required to avoid the + bugs with grabbing the wrong message off of the queue. This code + introduces a new interface to SMDI, with two dialplan functions. + First, you get an SMDI message in the dialplan using + SMDI_MSG_RETRIEVE() and then you access details in the message + using the SMDI_MSG() function. A side benefit of this is that it + now supports more than just chan_zap. For example, with this + implementation, you can have some FXO lines being terminated on a + SIP gateway, but the SMDI link in Asterisk. Another issue with + the current implementation is that it is quite common that the + station ID that comes in on the SMDI link is not necessarily the + same as the Asterisk voicemail box. There are now additional + directives in the smdi.conf configuration file which let you map + SMDI station IDs to Asterisk voicemail boxes. Yet another issue + with the current SMDI support was related to MWI reporting over + the SMDI link. The current code could only report a MWI change + when the change was made by someone calling into voicemail. If + the change was made by some other entity (such as with IMAP + storage, or with a web interface of some kind), then the MWI + change would never be sent. The SMDI module can now poll for MWI + changes if configured to do so. This work was inspired by and + primarily done for the University of Pennsylvania. (also related + to issue #9260) + +2008-02-26 00:03 +0000 [r104111] Jason Parker + + * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris. + (closes issue #12050) Reported by: asgaroth Patches: 12050.patch + uploaded by putnopvut (license 60) + +2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant + + * apps/app_chanspy.c: This patch fixes some pretty significant + problems with how app_chanspy handles pointers to channels that + are being spied upon. It was very likely that a crash would occur + if the channel being spied upon hung up. This was because the + current ast_channel handling _requires_ that the object is locked + or else it could disappear at any time (except in the owning + channel thread). So, this patch uses some channel datastore magic + on the spied upon channel to be able to detect if and when the + channel goes away. (closes issue #11877) (patch written by me, + but thanks to kpfleming for the idea, and to file for review) + + * main/utils.c: Improve the lock tracking code a bit so that a + bunch of old locks that threads failed to lock don't sit around + in the history. When a lock is first locked, this checks to see + if the last lock in the list was one that was failed to be + locked. If it is, then that was a lock that we're no longer + sitting in a trylock loop trying to lock, so just remove it. + (inspired by issue #11712) + +2008-02-25 21:37 +0000 [r104095] Joshua Colp + + * channels/chan_sip.c: Make it so a users.conf user creates both a + SIP peer and a SIP user. The user will be used for inbound + authentication for the device, and peer will be used for placing + calls to the device. (closes issue #9044) Reported by: queuetue + Patches: sip-gui-friend.diff uploaded by qwell (license 4) + +2008-02-25 21:31 +0000 [r104094] Tilghman Lesher + + * apps/app_voicemail.c: If the destination folder is full, don't + delete a message when exiting. (closes issue #12065) Reported by: + selsky Patch by: (myself) + +2008-02-25 20:49 +0000 [r104092] Jason Parker + + * main/config.c: Allow the use of #include and #exec in situations + where the max include depth was only 1. Specifically, this fixes + using #include and #exec in extconfig.conf. This was basically + caused because the config file itself raises the include level to + 1. I opted not to raise the include limit, because recursion here + could cause very bizarre behavior. Pointed out, and tested by + jmls (closes issue #12064) + +2008-02-25 18:38 +0000 [r104086] Russell Bryant + + * channels/chan_agent.c: Ensure that the channel doesn't disappear + in agent_logoff(). If it does, it could cause a crash. (fixes the + crash reported in BE-396) + +2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp + + * channels/chan_sip.c: If a resubscription comes in for a dialog we + no longer know about tell the remote side that the dialog does + not exist so they subscribe again using a new dialog. (closes + issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff + uploaded by file (license 11) + + * channels/chan_sip.c: Due to recent changes tag will no longer be + NULL if not present so we have to use ast_strlen_zero to see if + it's actually blank. (closes issue #12061) Reported by: flefoll + Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by + flefoll (license 244) + +2008-02-22 22:45 +0000 [r104037] Tilghman Lesher + + * channels/chan_sip.c: Backwards debug message. (closes issue + #12052) Reported by: flefoll Patches: + chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license + 244) + +2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson + + * channels/chan_zap.c: And as a followup to revision 104026, + completely remove event-related calls from a section of code + where we know there was no event to handle or get. + + * channels/chan_zap.c: Remove an incorrect debug message. It + reported that it had received a specific event and tried to + report which event was received. What actually was happening was + that it was reporting the number of bytes returned from a call to + read(). Thanks to Jared Smith for bringing the issue up on IRC + +2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming + + * main/manager.c: reduce the likelihood that HTTP Manager session + ids will consist of primarily '1' bits + +2008-02-20 22:32 +0000 [r103956] Mark Michelson + + * apps/app_queue.c: Clear up confusion when viewing the + QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the + user's perspective, the queue does exist, we shouldn't tell them + we couldn't find the queue. Instead since it is a dead queue, + report a 0 waiting count This issue was brought up on IRC by jmls + +2008-02-20 22:06 +0000 [r103953] Joshua Colp + + * channels/chan_zap.c: Don't wait for additional digits when + overlap dialing is enabled if the setup message contains the + sending_complete information element. (closes issue #11785) + Reported by: klaus3000 Patches: + sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by + klaus3000 (license 65) + +2008-02-20 21:40 +0000 [r103904] Mark Michelson + + * channels/chan_local.c: Fix a crash if the channel becomes NULL + while attempting to lock it. (closes issue #12039) Reported by: + danpwi + +2008-02-20 17:53 +0000 [r103845] Tilghman Lesher + + * main/stdtime/localtime.c: Compat fix for Solaris (closes issue + #12022) Reported by: asgaroth Patches: + 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) + Tested by: asgaroth + +2008-02-19 20:28 +0000 [r103823] Joshua Colp + + * channels/h323/ast_h323.cxx: Send CallerID Name in setup message. + (closes issue #11241) Reported by: tusar Patches: + h323id_as_callerid_name.patch uploaded by tusar (license 344) + +2008-02-19 20:02 +0000 [r103821] Russell Bryant + + * channels/chan_local.c: Account for the fact that the "other" + channel can disappear while the local pvt is not locked. (fixes a + problem introduced in rev 100581) (closes issue #12012) Reported + by: stevedavies Patch by me + +2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp + + * configure, configure.ac: Don't look for launchd when cross + compiling. (closes issue #12029) Reported by: ovi + + * channels/chan_sip.c: Fix building of chan_sip. + +2008-02-19 10:27 +0000 [r103806] Olle Johansson + + * channels/chan_sip.c: Make sure we send error replies correctly by + checking the via header. + +2008-02-18 23:56 +0000 [r103801] Joshua Colp + + * main/channel.c: Ensure that emulated DTMFs do not get interrupted + by another begin frame. (closes issue #11740) Reported by: gserra + Patches: v1-11740.patch uploaded by dimas (license 88) (closes + issue #11955) Reported by: tsearle (closes issue #10530) Reported + by: xmarksthespot + +2008-02-18 22:28 +0000 [r103790-103795] Jason Parker + + * channels/chan_zap.c: Fix previous commit so that we actually + disable echocanbridged if echocancel is off. + + * channels/chan_zap.c: Correct a message when echocancelwhenbridged + is on, but echocancel is not. Issue #12019 + +2008-02-18 20:52 +0000 [r103786] Mark Michelson + + * main/app.c: There was an invalid assumption when calculating the + duration of a file that the filestream in question was created + properly. Unfortunately this led to a segfault in the situation + where an unknown format was specified in voicemail.conf and a + voicemail was recorded. Now, we first check to be sure that the + stream was written correctly or else assume a zero duration. + (closes issue #12021) Reported by: jakep Tested by: putnopvut + +2008-02-18 17:31 +0000 [r103780] Tilghman Lesher + + * main/rtp.c, channels/chan_sip.c: When a SIP channel is being + auto-destroyed, it's possible for it to still be in bridge code. + When that happens, we crash. Delay the RTP destruction until the + bridge is ended. (closes issue #11960) Reported by: norman + Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 + (license 14) Tested by: norman + +2008-02-18 16:37 +0000 [r103770] Mark Michelson + + * channels/chan_zap.c: Fix a linked list corruption that under the + right circumstances could lead to a looped list, meaning it will + traverse forever. (closes issue #11818) Reported by: michael-fig + Patches: 11818.patch uploaded by putnopvut (license 60) Tested + by: michael-fig + +2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp + + * main/asterisk.c: Backport fix from issue #9325. (closes issue + #11980) Reported by: rbrunka + + * channels/chan_sip.c: Don't care if the extension given doesn't + exist for subscription based MWI. + +2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant + + * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race + condition (closes issue #11780) Reported by: guillecabeza + Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license + 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license + 380) + + * main/loader.c: In the case that you try to directly reload a + module has returned AST_MODULE_LOAD_DECLINE, log a message + indicating that the module is not fully initialized and must be + initialized using "module load". + + * main/loader.c: Don't attempt to execute the reload callback for a + module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash + that was reported against chan_console in trunk. (closes issue + #11953, reported by junky, fixed by me) + +2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson + + * doc/imapstorage.txt, configure, configure.ac: Final round of + changes for configure script logic for IMAP Now if a directory is + specified, then we will search that directory for a source + installation of the IMAP toolkit. If none is found, then we will + use that directory as the basis for detecting a package + installation of the IMAP c-client. If that check fails, then + configure will fail. + + * configure, configure.ac: Fix a bit of wrong logic in the + configure script that caused problems when trying to configure + without IMAP. Patch suggestion from phsultan, but I modified it + slightly. (closes issue #12003) Reported by: pj Tested by: + putnopvut + + * doc/imapstorage.txt, configure, configure.ac: I apparently + misunderstood one of the requirements of this configure change. + Now, if a source directory is specified with the --with-imap + option, and a valid source installation is not detected there, + then configure will fail and will not check for a package + installation. + + * doc/imapstorage.txt: Make a small clarification in the + documentation + + * doc/imapstorage.txt: Update documentation regarding configuration + of IMAP + + * apps/app_voicemail.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Change to the + configure logic regarding IMAP. Prior to this commit, if you + wished to configure Asterisk with IMAP support, you would use the + --with-imap configure switch in one of the following two ways: + --with-imap=/some/directory would look in the directory specified + for a UW IMAP source installation --with-imap would assume that + you had imap-2004g installed in .. relative to the Asterisk + source With this set of changes the two above options still work + the same, but there are two new behaviors, too. + --with-imap=system will assume that you have -libc-client.so + where you store your shared objects and will attempt to find + c-client headers in your include path either in the imap or + c-client directory. If either of the two original methods of + specifying the imap option should fail, then the check for + --with-imap =system will be performed in addition. It is only + after this "system" check that failure can happen. + + * apps/app_voicemail.c: Fix build for non-IMAP builds + + * apps/app_voicemail.c: Fix the new message count if delete=yes + when using IMAP storage. (closes issue #11406) Reported by: + jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license + 50) Tested by: jaroth + +2008-02-14 19:51 +0000 [r103683-103684] Jason Parker + + * funcs/func_cdr.c: swap location for this.. + + * funcs/func_cdr.c: Document the 'l' option to the CDR() function. + (Thanks voipgate for pointing out the option, and Leif for + providing text for it.) Closes issue #11695. + +2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher + + * channels/chan_agent.c: We aren't talking to ourselves; we're + talking to someone else. (closes issue #11771) Reported by: + msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982 + (license 22) Tested by: caio1982, msetim + + * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is + not loaded (which is a symbol dependency) (closes issue #11760) + Reported by: non-poster Patches: 20080114__bug11760.diff.txt + uploaded by Corydon76 (license 14) Tested by: Corydon76, + non-poster, jamesgolovich + +2008-02-12 22:24 +0000 [r103503-103504] Jason Parker + + * main/asterisk.c: revert accidental change from last commit. oops + + * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition + that was impossible. + +2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp + + * channels/chan_sip.c: Even if no CallerID name or number has been + provided by the remote party still use the configured sip.conf + ones. (closes issue #11977) Reported by: pj + + * apps/app_meetme.c: If entering a conference with the 'w' option + ensure that we can't listen or speak until the marked user + appears. (closes issue #11835) Reported by: alanmcmillan + +2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming + + * configs/zapata.conf.sample: improve 2BCT documentation a bit + (thanks Jared) + +2008-02-09 06:23 +0000 [r103197] Tilghman Lesher + + * apps/app_voicemail.c: Commit fix for being unable to send + voicemail from VoiceMailMain Reported by: William F Acker (via + the -users mailing list) Patch by: Corydon76 (license 14) + +2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson + + * apps/app_queue.c: Prevent a potential three-thread deadlock. Also + added a comment block to explicitly state the locking order + necessary inside app_queue. (closes issue #11862) Reported by: + flujan Patches: 11862.patch uploaded by putnopvut (license 60) + Tested by: flujan + + * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl + fails for Zaptel timing device. (closes issue #11891) Reported + by: tzafrir + +2008-02-08 15:08 +0000 [r102968] Joshua Colp + + * channels/chan_iax2.c: Make sure the presence of dbsecret is + factored into user scoring. (closes issue #11952) Reported by: + bbhoss + +2008-02-07 19:53 +0000 [r102858] Jason Parker + + * res/res_features.c: Specify which digit string was matched in + debug message. (closes issue #11949) Reported by: dimas Patches: + v1-feature-debug.patch uploaded by dimas (license 88) + +2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming + + * configs/zapata.conf.sample: document usage of 'transfer' + configuration option for ISDN PRI switch-side transfers + +2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp + + * channels/chan_sip.c: Only consider a T.38-only INVITE compatible + if we have both a joint capability between us and them and if + they provided T.38. + + * main/global_datastores.c: Add missing header file and + ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by: + snuffy + +2008-02-06 15:19 +0000 [r102651] Russell Bryant + + * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so + that it gets inherited by outbound channels. (due to a discussion + between me and a user via email) + +2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming + + * pbx/Makefile, res/Makefile: ensure that all remaining + multi-object modules are built using their proper CFLAGS and + include directory paths + +2008-02-06 00:26 +0000 [r102576] Tilghman Lesher + + * apps/app_voicemail.c: Move around some defines to unbreak ODBC + storage. (closes issue #11932) Reported by: snuffy + +2008-02-05 20:02 +0000 [r102453] Mark Michelson + + * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes + issue #11919) Reported by: eferro Patches: + mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) + Tested by: eferro + +2008-02-05 19:52 +0000 [r102450] Joshua Colp + + * channels/chan_sip.c: If a REGISTER attempt comes in that is a + retransmission of a previous REGISTER do not create a new nonce + value. (issue #BE-381) + +2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming + + * channels/Makefile: ensure that components of chan_misdn.so are + built using any special build options that the configure script + generated (reported by Philipp Kempgen on asterisk-dev) + +2008-02-05 15:09 +0000 [r102378] Joshua Colp + + * res/res_clioriginate.c: Perform dialing asynchronously when using + the originate CLI command so the CLI does not appear to block. + (closes issue #11927) Reported by: bbhoss + +2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher + + * main/asterisk.c, utils/muted.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Cross-platform + fix: OS X now deprecates the use of the daemon(3) API. (closes + issue #11908) Reported by: oej Patches: + 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76 + + * funcs/func_strings.c: Missing braces. (closes issue #11912) + Reported by: dimas Patches: sprintf.patch uploaded by dimas + (license 88) + +2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson + + * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE + (according to RFC 3261) (closes issue #9492) Reported by: + kryptolus Patches: bug9492.txt uploaded by oej (license 306) + Tested by: oej + + * channels/chan_sip.c: Handle ACK and CANCEL in an invite + transaction - even if we get INFO transactions during the actual + call setup. (closes issue #10567) Reported by: jacksch Tested by: + oej Patch by: oej inspired by suggestions from neutrino88 in the + bug tracker + +2008-02-01 23:06 +0000 [r101989] Russell Bryant + + * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns + out that even though G.722 is 16 kHz, it is supposed to specified + as 8 kHz in the RTP, and RTP timestamps are supposed to be + calculated based on 8 kHz. (Apparently this is due to a bug in a + spec, but people follow it anyway, because it's the spec ...) + +2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher + + * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded + voicemail, and fixed several other bugs while I'm in the area. + (closes issue #11615) Reported by: jamessan Patches: + 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, jamessan + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: Change detection of getifaddrs to use + AST_C_COMPILE_CHECK, backported from trunk (as suggested by + kpfleming) + +2008-02-01 17:41 +0000 [r101822] Jason Parker + + * apps/app_authenticate.c: Remove a needless (and incorrect) call + to feof() after fgets(). This would have exited the loop early if + you had an authentication file with no newline at the end. + +2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant + + * apps/app_authenticate.c: off by one error + + * apps/app_authenticate.c: Don't overwrite the last character of a + line if it's not a newline. This would happen if the last line in + the file doesn't have a newline. (pointed out by Qwell) + +2008-02-01 15:55 +0000 [r101772] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/acl.c: Compatibility fix for OpenWRT (reported by Brian + Capouch via the mailing list) + +2008-02-01 00:32 +0000 [r101693] Russell Bryant + + * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial + strings for the case that no peer or hostname was provided, which + is the one part of the dial string that is absolutely required. + If it's not there, bail out. (closes issue #11897) Reported by + sokhapkin Patch by me + +2008-02-01 00:06 +0000 [r101649] Mark Michelson + + * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle + periods of no channel activity" (closes issue #9256) Reported by: + cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt + uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, + rjain + +2008-01-31 Russell Bryant + + * Asterisk 1.4.18 released. + +2008-01-31 23:10 +0000 [r101601] Russell Bryant + + * main/translate.c, main/file.c: Fix a couple of places where + ast_frfree() was not called on a frame that came from a + translator. This showed itself by g729 decoders not getting + released. Since the flag inside the translator frame never got + unset by freeing the frame to indicate it was no longer in use, + the translators never got destroyed, and thus the g729 licenses + were not released. (closes issue #11892) Reported by: xrg + Patches: 11892.diff uploaded by russell (license 2) Tested by: + xrg, russell + +2008-01-31 21:00 +0000 [r101531] Mark Michelson + + * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the + beginning of an absolute pathname. 2. If ast_monitor_change_fname + is called and the new filename is the same as the old, then exit + early and don't set the filename_changed field in the monitor + structure. Setting it in this case was causing ast_monitor_stop + to erroneously delete them. (closes issue #11741) Reported by: + garlew Tested by: putnopvut + +2008-01-31 19:52 +0000 [r101482] Jason Parker + + * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes + for struct in_addr funkiness. Issue #11885, patch by snuffy. + +2008-01-31 19:30 +0000 [r101480] Steve Murphy + + * main/pbx.c: closes issue #11845; that's the one where there's a + 1004 byte cdr leak with every AMI Redirect to a zap channel + +2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant + + * channels/chan_agent.c: Add more missing locking of the agents + list ... + + * channels/chan_agent.c: Move the locking from find_agent() into + the agent dialplan function handler to ensure that the agent + doesn't disappear while we're looking at it. + + * channels/chan_agent.c: Add missing locking to the find_agent() + function. + +2008-01-30 15:41 +0000 [r101222] Joshua Colp + + * main/slinfactory.c: Fix an issue where if a frame of higher + sample size preceeded a frame of lower sample size and + ast_slinfactory_read was called with a sample size of the + combined values or higher a crash would happen. (closes issue + #11878) Reported by: stuarth + +2008-01-30 15:34 +0000 [r101219] Jason Parker + + * configs/extensions.conf.sample: Change default config to use + descending channel order of groups, rather than ascending. Fixes + a potential source of confusion in glare-type situations. Issue + 11875, reported by JimVanM. + +2008-01-30 15:23 +0000 [r101216] Mark Michelson + + * apps/app_queue.c: Fix a logic error with regards to autofill. + Prior to this change, it was possible for a caller to go out of + turn if autofill were enabled and callers ahead in the queue were + attempting to call a member. This change fixes this. + +2008-01-30 11:20 +0000 [r101152] Olle Johansson + + * channels/chan_sip.c: Stop musiconhold on attended transfer. + (closes issue #11872) Reported by: gareth Patches: + svn-101018.patch uploaded by gareth (license 208) + +2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard + + * build_tools/make_version: updated build_tools to handle the + autotag directory structure changes; changes related to BE-353. + Patch by The Russell and reviewed by The Me. + +2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson + + * apps/app_queue.c: Remove a memory leak from updating realtime + queues + + * apps/app_queue.c: Fixing an erroneous return value returned when + attempting to pause or unpause a queue member fails. Fixes + BE-366, thanks to John Bigelow for writing the patch. + +2008-01-29 17:57 +0000 [r100934] Joshua Colp + + * apps/app_mixmonitor.c: Don't forget to record the channel so we + know whether it is bridged or not later. (closes issue #11811) + Reported by: slavon + +2008-01-29 17:43 +0000 [r100932] Russell Bryant + + * main/Makefile: Fix the last couple of issues related to building + from a path that contains spaces. (closes issue #11834) + +2008-01-29 17:41 +0000 [r100930] Jason Parker + + * channels/misdn_config.c: Initialize an array to 0s if config + option not specified. (closes issue #11860) Patches: + misdn_get_config.v1.diff uploaded by IgorG (license 20) + +2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant + + * Makefile: Use GNU make magic instead of shell magic to escape + spaces in the working directory. (related to issue #11834) + + * Makefile: Fix building Asterisk when the working path has spaces + in it. (closes issue #11834) Reported by: spendergrass Patched + by: me + +2008-01-29 16:10 +0000 [r100835] Jason Parker + + * channels/chan_zap.c: Allow zap groups above 30 to work properly. + (closes issue #11590) Reported by: tbsky + +2008-01-29 10:36 +0000 [r100793] Christian Richter + + * channels/chan_misdn.c: fixed potential segfault in misdn show + channels CLI command + +2008-01-29 08:26 +0000 [r100740] Olle Johansson + + * channels/chan_sip.c: (closes issue #11736) Reported by: MVF + Patches: bug11736-2.diff uploaded by oej (license 306) Tested by: + oej, MVF, revolution (russellb: This was the showstopper for the + release.) + +2008-01-28 21:02 +0000 [r100675] Tilghman Lesher + + * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly + (went to 't' instead of 'T') + +2008-01-28 20:55 +0000 [r100673] Mark Michelson + + * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of + chan_vpb. It is alive and well. + +2008-01-28 20:42 +0000 [r100672] Jason Parker + + * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put + greeting files into the database like we do with the others. + Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded + by dimas (license 88) + +2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant + + * channels/chan_sip.c: For some reason, the use of this strdupa() + is leading to memory corruption on freebsd sparc64. This trivial + workaround fixes it. (closes issue #10300, closes issue #11857, + reported by mattias04 and Home-of-the-Brave) + + * res/res_features.c: Fix a crash in ast_masq_park_call() (issue + #11342) Reported by: DEA Patches: res_features-park.txt uploaded + by DEA (license 3) + +2008-01-28 18:23 +0000 [r100624] Jason Parker + + * channels/chan_zap.c: Correct a comment which made little/no + sense. + +2008-01-28 17:15 +0000 [r100581] Russell Bryant + + * main/channel.c, channels/chan_local.c, + include/asterisk/channel.h: Make some deadlock related fixes. + These bugs were discovered and reported internally at Digium by + Steve Pitts. - Fix up chan_local to ensure that the channel lock + is held before the local pvt lock. - Don't hold the channel lock + when executing the timing function, as it can cause a deadlock + when using chan_local. This actually changes the code back to be + how it was before the change for issue #10765. But, I added some + other locking that I think will prevent the problem reported + there, as well. + +2008-01-27 21:59 +0000 [r100465] Tilghman Lesher + + * main/rtp.c, channels/chan_mgcp.c, main/cdr.c, + channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c, + channels/chan_h323.c, include/asterisk/sched.h, main/file.c, + pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from + the scheduler, ignoring the return value could possibly cause + memory to be accessed after it is freed, which causes all sorts + of random memory corruption. Instead, if a deletion fails, wait a + bit and try again (noting that another thread could change our + taskid value). (closes issue #11386) Reported by: flujan Patches: + 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, flujan, stuarth` + +2008-01-25 22:32 +0000 [r100418] Mark Michelson + + * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is + now preferred that users of Voicetronix products use chan_zap in + combination with their zaptel drivers. + +2008-01-25 21:24 +0000 [r100378] Jason Parker + + * channels/chan_sip.c: This would have never been true, since we're + passing (sizeof(req.data) - 1) as the len to recvfrom(). + +2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming + + * include/asterisk/app.h: make these macros not assume that the + only other field in the structure is 'argc'... this is true when + someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable + to define your own structure as long as it has the right fields + +2008-01-24 17:22 +0000 [r100164] Russell Bryant + + * main/asterisk.c: Update main Asterisk copyright info to 2008 + +2008-01-24 16:41 +0000 [r100138] Jason Parker + + * main/acl.c: Fix compilation on Solaris. (closes issue #11832) + Patches: bug-11832.diff uploaded by snuffy (license 35) + +2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson + + * channels/chan_sip.c: Second attempt. Don't change invitestate + when receiving 18x messages in CANCEL state. (issue #11736) + Reported by: MVF Patch by oej. + + * channels/chan_sip.c: Make sure we don't cancel destruction on + calls in CANCEL state, even if we get 183 while waiting for + answer on our CANCEL. (issue #11736) Reported by: MVF Patches: + bug11736.txt uploaded by oej (license 306) Tested by: MVF + +2008-01-23 20:25 +0000 [r99975] Mark Michelson + + * apps/app_externalivr.c: Fixing a typo. + +2008-01-23 17:46 +0000 [r99923] Russell Bryant + + * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the + beginning of a list of channels to potentially spy on. However, + if there were no matching channels, it would beep at you over and + over, which is pretty annoying. Now, it will only beep once in + the case that there are no channels to spy on, but it will still + beep again once it reaches the beginning of the channel list + again. (closes issue #11738, patched by me) + +2008-01-23 16:18 +0000 [r99878] Mark Michelson + + * channels/chan_sip.c: These flag tests were illogical. They were + testing sip_peer flags on a sip_pvt. Thanks to Russell for + helping to get this odd problem figured out. + +2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher + + * apps/app_voicemail.c: When we reset the password via an external + command, we should also reset the password stored in the + in-memory list, too (otherwise it doesn't really take effect). + (closes issue #11809) Reported by: davetroy Patches: + fix_externpass.diff uploaded by davetroy (license 384) + + * res/res_odbc.c: Oops, should have checked for a NULL obj, here, + too + + * main/acl.c: Just confirmed that all current platforms need this + header file + +2008-01-22 20:56 +0000 [r99652] Olle Johansson + + * channels/chan_sip.c: Thanks to Russell's education I realize that + BUFSIZ has changed since I learned the C language over 20 years + ago... Resetting chan_sip to the size of BUFSIZ that I expected + in my old head to avoid to heavy memory allocations on some + systems. + +2008-01-22 20:34 +0000 [r99643] Tilghman Lesher + + * main/acl.c: Fix the defines for OS X (and Solaris, too) + +2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson + + * channels/chan_local.c, res/res_features.c, channels/chan_agent.c, + apps/app_followme.c: Add more dependencies on chan_local and add + a note to the description of chan_local so that people don't + disable it in menuselect just to clean up. + + * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial + still runs without chan_local, but will be missing forwarding + functionality. + +2008-01-22 16:54 +0000 [r99540] Tilghman Lesher + + * main/acl.c: Ensure that we can get an address even when we don't + have a default route. (closes issue #9225) Reported by: junky + Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 + (license 14) Tested by: oej, loloski, sergee + +2008-01-22 15:08 +0000 [r99501] Olle Johansson + + * channels/chan_sip.c: Cleaning up some documentation that led to + confusion in a bug report + +2008-01-21 23:55 +0000 [r99426] Mark Michelson + + * channels/chan_local.c: Fixing an issue wherein monitoring local + channels was not possible. During a channel masquerade, the + monitors on the two channels involved are swapped. In 99% of the + cases this results in the desired effect. However, if monitoring + a local channel, this caused the monitor which was on the local + channel to get moved onto a channel which is immediately hung up + after the masquerade has completed. By swapping the monitors + prior to the masquerade, we avoid the problem by tricking the + masquerade into placing the monitor back onto the channel where + we want it. During the investigation of the issue, the channel's + monitor was the only thing that was swapped in such a manner + which did not make sense to have done. All other variable + swapping made sense. + +2008-01-21 18:11 +0000 [r99341] Tilghman Lesher + + * res/res_odbc.c, configs/res_odbc.conf.sample, + include/asterisk/res_odbc.h: Permit the user to specify number of + seconds that a connection may remain idle, which fixes a crash on + reconnect with the MyODBC driver. (closes issue #11798) Reported + by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt + uploaded by Corydon76 (license 14) Tested by: mvanbaak + +2008-01-21 16:01 +0000 [r99301] Joshua Colp + + * channels/chan_sip.c: Bump the buffer size for Via headers up to + 512. There are some exceptionally large Via headers out there. + (closes issue #11783) Reported by: ofirroval + +2008-01-19 10:05 +0000 [r99187] Russell Bryant + + * main/slinfactory.c: Fix a couple of memory leaks with frame + handling. Specifically, ast_frame_free() needed to be called on + the frame that came from the translator to signed linear. + +2008-01-18 22:57 +0000 [r99127] Joshua Colp + + * include/asterisk/channel.h: Remove the __ in front of the unused + variable. This causes some compilers to freak out. + +2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant + + * include/asterisk/translate.h, main/frame.c: Revert adding the + packed attribute, as it really doesn't make sense why that would + do any good. Fix the real bug, which is to do the check to see if + the frame came from a translator at the beginning of + ast_frame_free(), instead of at the end. This ensures that it + always gets checked, even if none of the parts of the frame are + malloc'd, and also ensures that we aren't looking at free'd + memory in the case that it is a malloc'd frame. (closes issue + #11792, reported by explidous, patched by me) + + * include/asterisk/translate.h: Since we're relying on the offset + between the frame and the beginning of the translator pvt struct, + set the packed attribute to make sure we get to the right place. + (potential fix for issue #11792) + +2008-01-18 17:13 +0000 [r99032] Terry Wilson + + * res/res_features.c: This should at least temporarily fix a + problem where the 't' Dial option is incorrectly passed to the + transferee when built-in attended transfers are used. There is + still a problem with 'T', but better to fix some problems than no + problems while we work on it. (closes issue #7904) Reported by: + k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee + (license 138) Tested by: sergee, otherwiseguy + +2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni + + * configs/cdr.conf.sample: doh! revert a revert of a revert + (changed by mistake in 99010) + + * main/manager.c, configs/cdr.conf.sample: missed that one while + reverting + + * main/manager.c: reverting 99001 - We need the Max-Age for + extending the life of cookie mansession_id + +2008-01-17 22:37 +0000 [r99004] Russell Bryant + + * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: + Have IAX2 optimize the codec translation path just like chan_sip + does it. If the caller's codec is in our codec list, move it to + the top to avoid transcoding. (closes issue #10500) Reported by: + stevedavies Patches: iax-prefer-current-codec.patch uploaded by + stevedavies (license 184) iax-prefer-current-codec.1.4.patch + uploaded by stevedavies (license 184) Tested by: stevedavies, pj, + sheldonh + +2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming + + * main/manager.c: we should only send the Set-Cookie header to the + browser on the first response after creating a manager session, + not on every response (doing so causes the browser to clear any + local cookies it may have associated with the session) + +2008-01-17 16:19 +0000 [r98991] Jason Parker + + * configs/zapata.conf.sample: Add a clarification about the + immediate= option of zapata.conf Issue 11784, patch by klaus3000. + +2008-01-16 22:36 +0000 [r98982] Russell Bryant + + * .cleancount, include/asterisk/channel.h: Add an unused pointer to + the ast_channel struct. This makes the ast_channel structure + retain the same size as it had in previous 1.4 releases. Also, + all of the offsets for members in the structure are still the + same (except for the two pointers that got replaced for the new + spy/whisper architecture.) + +2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp + + * .cleancount: Bump up cleancount due to previous commit that + changed the channel structure. + + * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c, + main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h + (added), main/Makefile, include/asterisk/chanspy.h (removed), + include/asterisk/channel.h, main/audiohook.c (added): Replace + current spy architecture with backport of audiohooks. This should + take care of current known spy issues. + + * channels/chan_iax2.c: Add missing NULLs at end of two + ast_load_realtimes. (closes issue #11769) Reported by: tequ + Patches: chaniax.patch uploaded by dimas (license 88) + +2008-01-16 17:20 +0000 [r98964] Mark Michelson + + * channels/chan_local.c: Fix a deadlock in chan_local in + local_hangup. There was contention because the local_pvt was held + and it was attempting to lock a channel, which is the incorrect + locking order. (closes issue #11730) Reported by: UDI-Doug + Patches: 11730.patch uploaded by putnopvut (license 60) Tested + by: UDI-Doug + +2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp + + * main/dial.c: Introduce a lock into the dialing API that protects + it when destroying the structure. (closes issue #11687) Reported + by: callguy Patches: 11687.diff uploaded by file (license 11) + + * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes + issue #11777) Reported by: tootai + + * channels/chan_sip.c: Don't drop the old record route information + when dealing with packets related to a reinvite. (closes issue + #11545) Reported by: kebl0155 Patches: reinvite-patch.txt + uploaded by kebl0155 (license 356) + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, codecs/codec_speex.c, + configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later + versions use a separate library for some things so we need to use + it if present in codec_speex. (closes issue #11693) Reported by: + yzg + +2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant + + * channels/chan_sip.c: Change a buffer in check_auth() to be a + thread local dynamically allocated buffer, instead of a massive + buffer on the stack. This fixes a crash reported by Qwell due to + running out of stack space when building with LOW_MEMORY defined. + On a very related note, the usage of BUFSIZ in various places in + chan_sip is arbitrary and careless. BUFSIZ is a system specific + define. On my machine, it is 8192, but by definition (according + to google) could be as small as 256. So, this buffer in + check_auth was 16 kB. We don't even support SIP messages larger + than 4 kB! Further usage of this define should be avoided, unless + it is used in the proper context. + + * main/rtp.c, include/asterisk/translate.h, main/frame.c, + main/translate.c, main/abstract_jb.c, channels/chan_iax2.c, + codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for + some memory access errors pointed out by the valgrind2.txt output + on issue #11698. The issue here is that it is possible for an + instance of a translator to get destroyed while the frame + allocated as a part of the translator is still being processed. + Specifically, this is possible anywhere between a call to + ast_read() and ast_frame_free(), which is _a lot_ of places in + the code. The reason this happens is that the channel might get + masqueraded during this time. During a masquerade, existing + translation paths get destroyed. So, this patch fixes the issue + in an API and ABI compatible way. (This one is for you, + paravoid!) It changes an int in ast_frame to be used as flag + bits. The 1 bit is still used to indicate that the frame contains + timing information. Also, a second flag has been added to + indicate that the frame came from a translator. When a frame with + this flag gets released and has this flag, a function is called + in translate.c to let it know that this frame is doing being + processed. At this point, the flag gets cleared. Also, if the + translator was requested to be destroyed while its internal frame + still had this flag set, its destruction has been deffered until + it finds out that the frame is no longer being processed. + Admittedly, this feels like a hack. But, it does fix the issue, + and I was not able to think of a better solution ... + +2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp + + * channels/chan_sip.c: Based on the boundary found move over the + correct amount. (closes issue #11750) Reported by: tasker + + * channels/chan_sip.c: Accept "; boundary=" not just ";boundary=" + in the multipart mixed content type. (closes issue #11750) + Reported by: tasker + +2008-01-14 20:59 +0000 [r98849] Mark Michelson + + * apps/app_voicemail.c: Adding in appropriate unlocks for the locks + I added. Thanks to joetester on IRC for pointing this out. + +2008-01-14 17:38 +0000 [r98774] Russell Bryant + + * main/translate.c: Revert a change that introduces an unacceptable + performance hit and is causing memory leaks ... (from rev 97973) + +2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson + + * apps/app_queue.c: Fixing another compilation error. I'm a bit off + today :( + + * apps/app_queue.c: Oops. Last commit had compilation error. + + * apps/app_queue.c: Adding explicit defaults for missing options to + init_queue. This is necessary because if a user either removes or + comments one of these options and reloads their queues, the + option will not reset to its default, instead maintaining the + value from prior to the reload. Thanks to John Bigelow for + pointing this error out to me. + +2008-01-12 00:05 +0000 [r98467] Tilghman Lesher + + * res/res_odbc.c: Add a connection timeout attribute, as that was + what was intended with the login timeout, but ODBC divides it up + into 2 different timeouts. (Closes issue #11745) + +2008-01-11 22:46 +0000 [r98390] Russell Bryant + + * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems. + (closes issue #11646) Reported by: caio1982 Patches: + dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) + dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested + by: caio1982, mvanbaak + +2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni + + * main/http.c: Comment explaining how to force browser to always + read some html files from server. + +2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp + + * main/rtp.c: If the incoming RTP stream changes codec force the + bridge to break if the other side does not support it. (closes + issue #11729) Reported by: tsearle Patches: + new_codec_patch_udiff.patch uploaded by tsearle (license 373) + + * res/res_agi.c: If the channel is hungup during RECORD FILE send a + result code of -1 to be uniform with everything else. (closes + issue #11743) Reported by: davevg Patches: res_agi.diff uploaded + by davevg (license 209) + +2008-01-11 19:10 +0000 [r98315] Mark Michelson + + * main/channel.c: Properly report the hangup cause as no answer + when someone does not answer (closes issue #10574, reported by + boch, patched by moy) + +2008-01-11 18:25 +0000 [r98266] Tilghman Lesher + + * codecs/gsm/Makefile: Add another exception (which doesn't work) + for -march optimization flag. Reported by: thomasmebes Patch by: + tilghman (Closes issue #11563) + +2008-01-11 18:25 +0000 [r98265] Russell Bryant + + * doc/security.txt, main/asterisk.c, configure, + include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, + makeopts.in: Backport the ability to set the ToS bits on Linux + when not running as root. Normally, we would not backport + features into 1.4, but, I was convinced by the justification + supplied by the supplier of this patch. He pointed out that this + patch removes a requirement for running as root, thus reducing + the potential impacts of security issues. (closes issue #11742) + Reported by: paravoid Patches: libcap.diff uploaded by paravoid + (license 200) + +2008-01-11 17:22 +0000 [r98219] Joshua Colp + + * apps/app_followme.c: Ensure the return value of ast_bridge_call + is passed back up as the application return value. This is needed + for transfers to function so the PBX core knows to continue + execution. (closes issue #10327) Reported by: kkiely + +2008-01-11 15:52 +0000 [r98164] Tilghman Lesher + + * channels/chan_sip.c: Back out changes from revision 97077, since + it wasn't perfect + +2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant + + * main/frame.c: Fix samples vs. length calculations for g722 + + * main/translate.c: Simplify this code with a suggestion from Luigi + on the asterisk-dev list. Instead of using is16kHz(), implement a + format_rate() function. + + * main/translate.c: Fix various timing calculations that made + assumptions that the audio being processed was at a sample rate + of 8 kHz. + +2008-01-10 23:08 +0000 [r97973] Tilghman Lesher + + * channels/chan_sip.c, main/translate.c: 1) When we get a + translated frame out, clone it, because if the translator pvt is + freed before we use the frame, bad things happen. 2) Getting a + failure from ast_sched_delete means that the schedule ID is + currently running. Don't just ignore it. (Closes issue #11698) + +2008-01-10 21:57 +0000 [r97925] Mark Michelson + + * apps/app_voicemail.c: Let us leave a voicemail for ourself if we + have logged into VoiceMailMain and chosen to leave a message. + (closes issue #11735, reported and patched by jamessan) + +2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy + + * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the + same fixes for ael.flex as was done in 97849 for ast_expr2.fl; + overrode the normally generate yyfree func with our own version + that checks the pointer for non-null before passing to free(). + Also takes care of a little problem with 2.5.33 and the use of + the __STDC_VERSION__ macro. + + * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a + fix for 2 things: a problem Terry was having in OSX with null + pointers, which was my fault, as I probably forgot to run the sed + script last time I made mods. So, I moved the fix into the flex + input itself. Then, I found when I used flex 2.5.33, that it was + using __STDC_VERSION__, and that's not real good; so I added back + in a DIFFERENT sed script to fix that little mess. Tested + everything, a couple different ways. Hope I did no harm, at the + least. + +2008-01-10 20:12 +0000 [r97847] Jason Parker + + * include/asterisk/frame.h: Fix a comment that is no longer true. + +2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant + + * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample, + pbx/kdeconsole_main.cc (removed): Remove other remnants of + pbx_kdeconsole + + * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in, + configure, include/asterisk/autoconfig.h.in, configure.ac, + makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't + worked in ages, and nobody has complained. (closes issue #11706, + reported by caio1982) + +2008-01-10 15:07 +0000 [r97697] Joshua Colp + + * funcs/func_groupcount.c: Don't try to copy the category from the + group if no category exists. (closes issue #11724) Reported by: + IgorG Patches: group_count.v1.patch uploaded by IgorG (license + 20) + +2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant + + * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose + messages + + * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't + actually load on my system still (related to issue #11706) + +2008-01-09 20:28 +0000 [r97618-97622] Jason Parker + + * main/cli.c: Correctly display a message if a command could not be + found. Also fix a comment which may have led to this happening. + Issue 11718, reported by kshumard. + + * main/cli.c: Fix some locking and return value funkiness. We + really shouldn't be unlocking this lock inside of a function, + unless we locked it there too. + +2008-01-09 18:48 +0000 [r97575] Mark Michelson + + * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some + smaller functions this time + +2008-01-09 18:02 +0000 [r97529] Russell Bryant + + * res/res_features.c: Fix saying the parking space number to the + caller doing the parking ... + +2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming + + * codecs/codec_zap.c: report the same message whether Zaptel does + not have transcoder support loaded or no transcoders were found + +2008-01-09 16:44 +0000 [r97489] Philippe Sultan + + * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc + function. As underlined in issue #10437 by Josh, we need to + prevent a possible memory leak. We only set the name part of the + caller id, the number part is not relevant when dealing with + JIDs. Closes issue #11549. + +2008-01-09 16:11 +0000 [r97450] Joshua Colp + + * apps/app_meetme.c: Don't do conferencing totally in Zaptel if + Monitor is running on the channel. (closes issue #11709) Reported + by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy + (license 371) + +2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming + + * channels/chan_zap.c: pass the right variable to get an error + string... oops + + * channels/chan_zap.c: add error number output to ioctl failure + messages to help with debugging + +2008-01-09 00:44 +0000 [r97350] Tilghman Lesher + + * main/cli.c, main/editline/readline.c: Allow filename completion + on zero-length modules, remove a memory leak, remove a file + descriptor leak, and make filename completion thread-safe. + Patched and tested by tilghman. (Closes issue #11681) + +2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson + + * apps/app_queue.c: use the \retval doxygen command properly + + * apps/app_queue.c: Part 1 of N of adding doxygen comments to + app_queue. I picked some of the most common functions used (which + also happen to be some the biggest/ugliest functions too) to + document first. I'm pretty new to doxygen so criticism is + welcome. + + * apps/app_queue.c: Some coding guidelines-related cleanup + +2008-01-08 20:48 +0000 [r97195] Joshua Colp + + * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp. + (closes issue #11443) Reported by: eferro Patches: + dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license + 337) + +2008-01-08 20:47 +0000 [r97194] Tilghman Lesher + + * main/autoservice.c, main/utils.c: Increase constants to where + we're less likely to hit them while debugging. (Closes issue + #11694) + +2008-01-08 20:42 +0000 [r97192] Mark Michelson + + * apps/app_voicemail.c: Making some changes designed to not allow + for a corrupted mailstream for a vm_state. 1. Add locking to the + vm_state retrieval functions so that no linked list corruption + occurs. 2. Make sure to always grab the persistent vm_state when + mailstream access is necessary. 3. Correct an incorrect return + value in the init_mailstream function. (closes issue #11304, + reported by dwhite) + +2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp + + * funcs/func_groupcount.c: If no group has been provided to the + GROUP_COUNT dialplan function then use the first one specific to + the channel. (closes issue #11077) Reported by: m4him + + * apps/app_queue.c: Make app_queue calls work with directed pickup. + (closes issue #11700) Reported by: jbauer + +2008-01-08 18:02 +0000 [r97077] Tilghman Lesher + + * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes, + found in issue #11386, but not completely closing that issue. + +2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant + + * configs/extensions.conf.sample, /: Merged revisions 96931 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | + 2 lines Change misery.digium.com to pbx.digium.com ........ + + * res/res_smdi.c: Don't crash if something happens when setting up + an SMDI interface and it gets destroyed before the SMDI port + handling thread gets created. + +2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan + + * res/res_jabber.c: Indentation fix, makes the code easier to read + + * res/res_jabber.c: Compute the base64 value over the + [authzid]\0authcid\0password string, thus excluding the trailing + NULL byte. This change has already been committed to trunk, see + #11644. + +2008-01-05 02:09 +0000 [r96644] Russell Bryant + + * main/devicestate.c: Don't pass an empty string as the device + name. + +2008-01-04 23:03 +0000 [r96575] Tilghman Lesher + + * main/devicestate.c: Fix the problem of notification of a device + state change to a device with a '-' in the name. Could probably + do with a better fix in trunk, but this bug has been open way too + long without a better solution. Reported by: stevedavies Patch + by: tilghman (Closes issue #9668) + +2008-01-04 22:55 +0000 [r96573] Jason Parker + + * res/res_features.c: Properly continue in the dialplan if using + PARKINGEXTEN and the slot is full. Issue 11237, patch by me. + +2008-01-04 19:27 +0000 [r96525] Tilghman Lesher + + * channels/chan_sip.c: If you change the bindaddr in sip.conf to a + non-bound address and reload, sip goes kablooie. Reported and + patched by: one47 (Closes issue #11535) + +2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant + + * channels/chan_zap.c: Make use of the temporary channel pointer + while the pvt is unlocked. (closes issue #11675) Reported by: + flefoll Patches: chan_zap.c.patch-store-owner-before-unlock + uploaded by flefoll (license 244) + + * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has + been destroyed before we get to this point (closes issue #11672, + reported by snuffy, patched by me) + +2008-01-03 21:37 +0000 [r96318] Tilghman Lesher + + * res/res_config_pgsql.c: Missed initialization caused crash. + Reported and fixed by: tiziano (Closes issue #11671) + +2008-01-03 12:12 +0000 [r96198-96199] Christian Richter + + * channels/chan_misdn.c: make sure frame is completely clean, + before we send it to asterisk as DTMF. If we don't make it clean, + it happens that one way audio occurs.. + + * channels/chan_misdn.c: when overlapdial was used and no number + was dialed, the call was dropped, now we just jump into the s + extension, which makes a lot more sense. + +2008-01-02 23:46 +0000 [r96102] Mark Michelson + + * apps/app_queue.c: We need to reset the membername to NULL on each + iteration of this loop, otherwise the result is that multiple + members can have the same name, since the variable was not reset + on each iteration of the loop. + +2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant + + * pbx/pbx_config.c: Convert locks of the contexts list in + pbx_config to the appropriate rdlock or wrlock + + * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts + list. + + * apps/app_macro.c: app_macro only needs a rdlock on the contexts + list. + +2008-01-02 Russell Bryant + + * Asterisk 1.4.17 released. + +2008-01-02 20:24 +0000 [r95946] Joshua Colp + + * channels/chan_sip.c: Allocate a SIP refer structure when + performing a transfer using BYE with Also so that the transfer + information is properly stored. (AST-2008-028) (closes issue + #11637) Reported by: greyvoip + +2008-01-02 17:51 +0000 [r95890] Mark Michelson + + * apps/app_queue.c: A change to improve the accuracy of queue + logging in the case where a member does not answer during the + specified timeout period. Prior to this change, there was a small + chance that the member name recorded in this case would be blank. + Also prior to this change, if using the ringall strategy, if no + one answered the call during the specified timeout, the member + name listed in the queue log would randomly be one of the members + that was rung. (closes issue #11498, reported and tested by + hloubser, patched by me) + +2007-12-31 23:43 +0000 [r95577] Mark Michelson + + * main/pbx.c: Avoiding a potentially bad locking situation. + ast_merge_contexts_and_delete writelocks the conlock, then calls + ast_hint_extension, which attempts to readlock the same lock. + Recursion with read-write locks is dangerous, so the inner lock + needs to be removed. I did this by copying the "guts" of + ast_hint_extension into ast_merge_contexts_and_delete (sans the + extra lock). (this change is inspired by the locking problems + seen in issue #11080, but I have no idea if this is the + problematic area experienced by the reporters of that issue) + +2007-12-31 20:27 +0000 [r95470] Tilghman Lesher + + * funcs/func_env.c: Allow the default "0" to be returned if the + STAT fails (Closes issue #11659) + +2007-12-28 18:24 +0000 [r95191] Russell Bryant + + * channels/chan_sip.c: Remove duplicate increment of the header + count in the add_header() function. (closes issue #11648) + Reported by: makoto Patch provided by sergee, committed patch by + me, inspired by comments from putnopvut + +2007-12-28 00:16 +0000 [r95095] Mark Michelson + + * apps/app_queue.c: I found a bug while browsing the queue code and + managed to reproduce it in a small setup. If a queue uses the + ringall strategy, it was possible through unfortunate coincidence + for a single member at a given penalty level to make app_queue + think that all members at that penalty level were unavailable and + cause the members at the next penalty level to be rung. With this + patch, we will only move to the next penalty level if ALL the + members at a given penalty level are unreachable. + +2007-12-27 21:40 +0000 [r95024] Russell Bryant + + * main/channel.c: Don't report a syntax error when an empty string + is passed to ast_get_group. Just return 0. (closes issue #11540) + Reported by: tzafrir Patches: group_empty.diff uploaded by + tzafrir (license 46) -- slightly changed by me + +2007-12-27 20:09 +0000 [r94977] Mark Michelson + + * main/io.c: Fixing a typo in a comment. + +2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp + + * channels/chan_h323.c: Include types.h in chan_h323 as without it + it can not be compiled on some operating systems like FreeBSD to + name one. (closes issue #11585) Reported by: sobomax Patches: + chan_h323.c.diff uploaded by sobomax (license 359) + + * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is + set or not on the dialog. It is possible for it to be a pointer + to NULL. (closes issue #11557) Reported by: FuriousGeorge + +2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant + + * main/pbx.c: Now that the contexts lock is a read/write lock, it + should not be locked here in ast_hint_state_changed(). This makes + it get locked recursively which now causes a deadlock. (closes + issue #11080, thanks to callguy for the access to a deadlocked + machine) + + * include/asterisk/translate.h, main/translate.c: Use the constant + that I really meant to use here ... + + * main/translate.c: Change ast_translator_best_choice() to only pay + attention to audio formats. This fixes a problem where Asterisk + claims that a translation path can not be found for channels + involving video. (closes issue #11638) Reported by: cwhuang + Tested by: cwhuang Patch suggested by cwhuang, with some + additional changes by me. + +2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming + + * main/manager.c: make this comment explain the situation in an + even more explicit fashion + +2007-12-26 20:43 +0000 [r94808] Tilghman Lesher + + * main/manager.c: Workaround for what is probably a glibc bug (but + we'll see this crop up again and again, if we don't add the + workaround). Reported by: rolek Patch by: tilghman (Closes issue + #11601, closes issue #11426) + +2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant + + * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag + was already set before starting autoservice, remember it and + ensure that the channel has the same setting when autoservice + gets stopped. (pointed out by d1mas, patched up by me) + + * main/autoservice.c: When a channel is in autoservice, mark a flag + on the channel that says that we only care about the END of a + digit. That way, no magic digit emulation stuff will happen when + all we're doing is queueing up END frames. + + * res/res_features.c: Don't try to send a parked call back to + itself. (closes issue #11622, reported by djrodman, patched by + me) + + * main/autoservice.c: Don't store DTMF BEGIN frames while a channel + is in autoservice. It's just going to make ast_read() do a lot of + extra work when the channel comes back out of autoservice. + (closes issue #11628, patched by me) + + * Makefile: List include/asterisk/version.h as a .PHONY target + because we want the commands listed for this target to be + executed regardless of whether the file exists or not. This fixes + having the version not up to date when running from svn. (closes + issue #11619, reported by plack, fixed by me) + +2007-12-25 02:27 +0000 [r94769] Joshua Colp + + * channels/chan_sip.c: file says... build on the builders. + +2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher + + * main/channel.c: Race: we need to wait to queue a NewChannel event + until after the channel is inserted into the channel list. The + reason is because some manager users immediately queue requests + from the channel when they see that event and are confused when + Asterisk reports no such channel. (Closes issue #11632) + + * channels/chan_sip.c: More deadlock avoidance code (this time + between sip_monitor and sip_hangup) Reported by: apsaras Patch + by: tilghman (Closes issue #11413) + + * channels/chan_sip.c: Another bit of bad logic in realtime_peer + Reported by: dimas Patch by: dimas (Closes issue #11631) + +2007-12-23 01:21 +0000 [r94660] Tilghman Lesher + + * channels/chan_sip.c: Argh... I suppose third time's the charm. + +2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson + + * apps/app_voicemail.c: Bunch of coding guidelines cleanup + + * apps/app_voicemail.c: Better quota support for using IMAP storage + voicemail (closes issue #11415, reported by jaroth) (closes issue + #11152, reported by selsky) Patch provided by jaroth + + * apps/app_voicemail.c: The mail_copy c-client function does not + expect a full imap mailbox string, just the name of the mailbox. + (closes issue #11419, reported and patched by jaroth, with + additional patchwork from me) + + * main/dial.c: Since we are freeing list elements within a list + traversal, we need to use the safe traversal and remove the item + from the list before freeing it. (closes issue 11612, reported by + dtyoo) + +2007-12-21 16:37 +0000 [r94466] Russell Bryant + + * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to + a read/write lock to resolve a deadlock. This has a nice side + benefit of improving performance. :) (closes issue #11609) + (closes issue #11080) + +2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson + + * apps/app_queue.c: Removing a debug message I accidentally just + committed + + * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying + dates and times. Also some coding guidelines cleanup. (closes + issue #11599, reported and patched by caio1982, coding guidelines + cleanup by me) + +2007-12-21 15:07 +0000 [r94418] Tilghman Lesher + + * main/asterisk.c: Fix for restart-as-user problem reported via the + -dev list + +2007-12-20 Russell Bryant + + * Asterisk 1.4.16.2 released. + +2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) | + 5 lines Fix another potential seg fault ... (closes issue #11606) + Reported by: dimas ........ + + * channels/chan_zap.c: Fix a deadlock in d-channel handling in + chan_zap. This deadlock was introduced by the fix to ensure that + channels are properly locked when handling channel variables. + There were sections of this code where the channel pvt was locked + before the channel lock, when in fact it _must_ be the other way + around. (closes issue #11582) Reported by: bugi + +2007-12-19 23:02 +0000 [r94122] Mark Michelson + + * res/res_monitor.c: Sox versions 13.0.0 and newer do not have + "soxmix" and instead use sox -m. res_monitor needs to use this if + the user does not have soxmix. (closes issue #11589, reported by + amessina, patch inspired by amessina but with a flourish from me) + +2007-12-19 22:48 +0000 [r94077] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Check + for the existence of the soxmix application on the target + platform and have the result available in autoconfig.h. (part of + issue #11589) + +2007-12-19 Russell Bryant + + * Asterisk 1.4.16.1 released. + +2007-12-19 17:29 +0000 [r93955] Joshua Colp + + * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is + NULL. + +2007-12-19 17:04 +0000 [r93949] Tilghman Lesher + + * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't + defined (Closes issue #11602) + +2007-12-18 22:42 +0000 [r93764] Jason Parker + + * channels/chan_skinny.c: FreeBSD also does not have byte swap + functions. Issue 11586, patch by sobomax. + +2007-12-18 Russell Bryant + + * Asterisk 1.4.16 released. + +2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher + + * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions + 93667 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) + | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ + +2007-12-18 17:02 +0000 [r93625] Mark Michelson + + * main/channel.c: Rework deadlock avoidance used in ast_write, + since it meant that agent channels which were being monitored had + one audio file recorded and one empty audio file saved. (closes + issue #11529, reported by atis patched by me) + +2007-12-17 22:56 +0000 [r93381-93420] Jason Parker + + * main/translate.c: What was I thinking when I wrote this + masterpiece? -1 + 1 = 0.. who woulda thunk it?. + +2007-12-17 22:28 +0000 [r93377] Joshua Colp + + * main/utils.c: Do not try to access information about a lock when + printing out a trylock attempt. It is possible for the lock that + it references to no longer be valid. This would have caused + segfaults or deadlocks. (issue #BE-263) (closes issue #11080) + Reported by: callguy (closes issue #11100) Reported by: callguy + +2007-12-17 21:12 +0000 [r93336] Tilghman Lesher + + * include/asterisk/time.h: Today is tomorrow's yesterday, and + yesterday's tomorrow is today, and tomorrow's tomorrow is the day + after tomorrow, so who cares if you recycle anyway? If this + confuses you, that's nothing compared to what this fixes. ;-) + +2007-12-17 19:53 +0000 [r93291] Mark Michelson + + * apps/app_voicemail.c: We need to create the directory for a + voicemail user even if they are using IMAP storage since + greetings are stored in the filesystem. (closes issue #11388, + reported by spditner, patch by me inspired by a patch by + spditner) + +2007-12-17 18:05 +0000 [r93250] Joshua Colp + + * channels/chan_zap.c: If a call is received with a called number + IE containing nothing go to the 's' extension. (closes issue + #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt + uploaded by Corydon76 (license 14) + +2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming + + * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile, + res/Makefile, channels/Makefile, formats/Makefile: fix some + copy-and-paste leftovers + +2007-12-17 07:15 +0000 [r93182] Olle Johansson + + * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c, + apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add + dependencies on res_monitor and res_features. I wonder if + Asterisk can run at all without res_features. My guess is that + there's propably a lot of more modules and the core that depends + on it. Reported by: caio1982 (closes issue #11574) + +2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming + + * formats, Makefile, codecs/Makefile, funcs, apps/Makefile, + configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in, + formats/Makefile, pbx, res, channels, funcs/Makefile, codecs, + include/asterisk/autoconfig.h.in, build_tools/make_version, apps, + configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps + (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In + http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, + rizzo brought up some issues related to the way that the metadata + required for menuselect and the rest of the build system is + extracted from the source files. Since I had a few hours to kill + on an airplane today, I decided to improve this situation... so + now the system caches the extracted metadata and uses it to build + the menuselect 'tree' as much as it can. The result of this is + that when a single source file is changed, only the metadata for + that file needs to be extracted again, and the rest is used from + the cache files. I also reduced the number of forked processes + required to do the metadata extraction; it was actually possible + to do most of what we needed in the Makefiles themselves without + using any shell scripts at all! On my laptop, these changes + resulted in an 80% decrease in the time required for the + 'menuselect.makeopts' automatic check to occur after editing a + single source file. While doing this work I also cleaned up a few + minor things in the Makefiles, adding a check for 'awk' to the + configure script and changed all remaining places we use 'grep' + or 'awk' to use the ones found by the configure script, and + changed the 'prep_tarball' script to build the menuselect + metadata so that tarballs of Asterisk will include it and won't + require the user to wait while it is extracted after unpacking. + +2007-12-14 17:36 +0000 [r93000] Russell Bryant + + * main/config.c: There are a lot of existing systems that #include + non-existent files. So, to make the transition to treating this + as an error a bit less painless, just issue a huge error message + for now. Then, later, we can reinstate the code that treats it as + a failure. (Thanks to philippel for the feedback) + +2007-12-14 15:16 +0000 [r92937] Joshua Colp + + * channels/chan_sip.c: Up the length of the format on the SIP + channel since it can now be rather long. (closes issue #11552) + Reported by: francesco_r + +2007-12-14 15:05 +0000 [r92934] Christian Richter + + * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS + state setting and overlap_task thread starting. + +2007-12-14 15:01 +0000 [r92933] Tilghman Lesher + + * res/res_agi.c: Change help documentation to match actual behavior + (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman + (Closes issue #11548) + +2007-12-14 01:24 +0000 [r92875] Mark Michelson + + * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS, + don't spam the CLI with messages about possible deadlocks. + Instead just print the intended single message every five + seconds. (closes issue 11537, reported and patched by dimas) + +2007-12-13 21:28 +0000 [r92815] Tilghman Lesher + + * channels/chan_zap.c: Properly initialize polarity statuses, so + that they are detected properly. Reported by: julianjm Patch by: + julianjm (Closes issue #10238) + +2007-12-13 20:13 +0000 [r92809] Jason Parker + + * main/pbx.c: Make application help text a little more clear about + the use of extensions in a filename. + +2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson + + * apps/app_voicemail.c: Prevent another potential fd leak + + * apps/app_voicemail.c: Prevent a possible fd leak. + +2007-12-13 00:11 +0000 [r92696] Jason Parker + + * main/config.c, channels/chan_sip.c, channels/chan_h323.c, + channels/chan_iax2.c: If a typo is found in a config file, we + previous continued on with what was already loaded. We do not + want to do this (see bug below for details). This makes it so + that if a [ is found without a ], the entire config will fail, + and nothing in it will be loaded. Isue #10690. + +2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming + + * codecs/codec_zap.c: emit a warning message when we drop a G.729B + CNG frame destined for the transcoder + +2007-12-12 21:15 +0000 [r92617] Jason Parker + + * apps/app_meetme.c: Don't increment user count until after name + has been recorded (if enabled). Issue 11048, tested by pep. + +2007-12-12 19:40 +0000 [r92556] Russell Bryant + + * res/res_features.c: resolve compiler warning + +2007-12-12 17:46 +0000 [r92510] Mark Michelson + + * res/res_features.c: Correctly detect where a dynamic feature was + activated. Before this patch, the channel which initiated the + bridge was always assumed to have been the one which activated + the dynamic feature. This patch corrects this. (closes issue + #11529, reported and patched by nic_bellamy) + +2007-12-12 16:52 +0000 [r92463] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Test + directly for the API that fixed AST-2007-026, to ensure that + older versions of PostgreSQL are no longer acceptable. (Closes + issue #11526) + +2007-12-12 16:08 +0000 [r92443] Mark Michelson + + * apps/app_queue.c: Removing an unused variable. + +2007-12-11 19:51 +0000 [r92363] Joshua Colp + + * main/global_datastores.c: Fix potential memory leak with the + dialed interfaces list if another memory allocation fails. + (closes issue #11507) Reported by: eliel Patches: + global_datastores.c.patch uploaded by eliel (license 64) + +2007-12-11 17:42 +0000 [r92323] Mark Michelson + + * apps/app_queue.c: Fixing autofill to be more accurate. + Specifically, if calls ahead of the current caller were ringing + members (but not yet bridged) there could be available members + and waiting callers who would not get matched up. The member + availability checker was correctly determining the number of + available members in this scenario, but the queue itself did not + parallelly reflect this status on the pending calls. This commit + corrects the issue. (closes issue #11459, reported by + equissoftware, patched by me) + +2007-12-10 16:36 +0000 [r92204] Joshua Colp + + * main/rtp.c: Add G729A as another possible payload name for G729. + Some devices use this instead of G729, which is perfectly normal + since the payload number itself is defined and can't be used by + anything else so the name doesn't matter that much. (closes issue + #11483) Reported by: revolution Patches: rtp.diff uploaded by + revolution (license 346) + +2007-12-10 16:29 +0000 [r92202] Mark Michelson + + * apps/app_queue.c: If there are no members in a queue, then the + loop where the datastore for detecting duplicate dialed numbers + will be skipped, meaning the datastore isn't created. This means + that when we try to free it, there's a crash. This stops that + crash from occurring. (closes issue #11499, reported by slavon, + patched by eliel) + +2007-12-10 16:13 +0000 [r92200] Joshua Colp + + * channels/chan_sip.c: It is possible for nativeformats to contain + more then one codec, so print out multiple ones. (closes issue + #11366) Reported by: ovi + +2007-12-10 14:04 +0000 [r92158] Olle Johansson + + * channels/chan_sip.c: Avoid reinvite race situations with two + Asterisks trying to reinvite each other in 1.4 and trunk. This + patch implements support for the 491 error code that Asterisk 1.4 + generates on situations where we get an incoming INVITE and + already has one in progress. Thanks to mavetju for reporting and + to Raj Jain for an excellent explanation of the problem. Patch by + myself. Tested with 8 Asterisk servers connected to each other in + a training network. Closes issue #10481 + +2007-12-07 23:29 +0000 [r91890] Jason Parker + + * main/dsp.c: We need to make sure we free the input frame if we + return a different frame in ast_dsp_process. Issue 11273, pointed + out by dimas, with a patch by eliel. + +2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming + + * codecs/codec_zap.c: even though Asterisk explicitly requests that + endpoints using G.729 do *not* use Annex B (silence detection and + comfort noise generation) some do anyway; the transcoder card + interface does not currently work properly with CNG frames, so + trim off the CNG before sending the data + +2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant + + * main/utils.c: Make the lock protecting each thread's list of + locks it currently holds recursive. I think that this will fix + the situation where some people have said that "core show locks" + locks up the CLI. (related to issue #11080) + + * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS + code. The ast_mutex_init() function had the mutex attribute + object marked as static. This means that multiple threads + initializing locks at the same time could step on each other and + end up with improperly initialized locks. (found when tracking + down locking issues related to issue #11080) + + * include/asterisk/lock.h: I love fixing lock related errors in the + lock debugging code. That's about as ironic as it gets in + Asterisk programming land. Anyway, I spotted this bug while + trying to track down why systems are locking up and acting weird + in issue #11080. The mutex attribute object was marked as static + in this function when it should not have been. + + * apps/app_dial.c: * Add channel locking around datastore + operations that expect the channel to be locked. * Document why + we don't record Local channels in the dialed interfaces list. * + Remove the dialed variable as it isn't needed. * Restructure some + code for clarity and coding guidelines stuff + + * apps/app_queue.c: * Add channel locking around datastore + operations that expect the channel to be locked. * Document why + we don't record Local channels in the dialed interfaces list. * + Handle memory allocation failure. * Remove the dialed variable, + as it wasn't actually needed. * Tweak some formatting to conform + to coding guidelines. + + * main/autoservice.c: * Add a bit more of a verbose comment as to + why a hangup frame needs to be queued up if autoservice gets a + NULL return from ast_read(). * Make the process of queueing the + hangup frame more efficient by putting the frame where it is + going to end up and avoiding some locking and extra memory + allocations and freeing. + +2007-12-07 15:39 +0000 [r91737] Mark Michelson + + * main/autoservice.c: Hangups that happen during autoservice were + not processed appropriately. This is because a hangup actually + causes a NULL frame to be received, not a hangup frame. Queueing + a hangup if we receive a NULL frame during autoservice corrects + this problem (closes issue #11467, reported by jmls, patched by + me) + +2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant + + * apps/app_dial.c: Don't unlock the dialed_interfaces list until + we're done messing with the iterator. + + * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels + from Queue() and Dial() again. There was a slight flaw in the + code to prevent call forwards from looping that caused this + problem. (related to issue #11486) + + * apps/app_queue.c: Fix in an issue in the call forwarding handling + code that was causing crashes on every call into a queue. I'm not + entirely sure about the logic in this part of the code, so I want + to look at it some more tomorrow. However, this makes it safe and + keeps it from crashing. (closes issue #11486, reported by adamg, + patched by me) + +2007-12-07 00:52 +0000 [r91637] Tilghman Lesher + + * main/rtp.c: At the end of a call, when we're reporting, RTCP may + already be partially torn down, so check for NULL dereference + Reported by: blitzrage Patch by: tilghman (Closes issue #11450) + +2007-12-06 20:25 +0000 [r91541] Mark Michelson + + * apps/app_voicemail.c: IMAP storage did not honor the maxmsg + setting in voicemail.conf, and it also had the possibility of + crashing if a user had more than 256 messages in their voicemail. + This patch kills two birds with one stone by adding maxmsg + support and also setting a hard limit on the number of messages + at 255 so that the crashes cannot happen. (closes issue #11101, + reported by Skavin, patched by me) + +2007-12-06 19:11 +0000 [r91501] Russell Bryant + + * main/loader.c, include/asterisk/module.h: Add a new module flag + to indicate that a build sum is present. Modules built against + older Asterisk 1.4 headers will now load properly with just a + warning indicating that they are old and may cause problems. + (patch by paravoid) + +2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp + + * main/udptl.c: Fix various in the udptl implementation. It could + return empty modem frames, have an incorrect sequence number on + packets, and display the wrong sequence number in the debug + messages. (closes issue #11228) Reported by: Cache Patches: + udptl-4.patch uploaded by dimas (license 88) + + * channels/chan_sip.c: Add support for accepting and sending T.38 + in the initial INVITE. (closes issue #9402) Reported by: thdei + +2007-12-06 12:54 +0000 [r91366] Olle Johansson + + * main/loader.c, include/asterisk/logger.h, main/logger.c: Make + sure logger is reloaded at general reload in the cli. (Discovered + during Asterisk training in Portugal) + +2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson + + * apps/app_voicemail.c: Reverting extra stuff I didn't mean to + commit + + * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial() + did not properly handle the case where only a label was provided. + This was due to the fact that the answering channel did not have + an extension set, so ast_parseable_goto would fail. This fix + eliminates the call to ast_parseable_goto on the answering + channel since it is a wasteful call. The answering channel and + the calling channel are both directed to the same extension and + context, just different priorities, so we can just copy the + values from the calling channel to the answering channel and + increment the answering channel's priority. (closes issue #11382, + reported by jon, patch by me with correction by jon) + +2007-12-05 21:38 +0000 [r91237] Tilghman Lesher + + * sounds/Makefile: Upgrade to the latest version of extra sounds + +2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant + + * main/threadstorage.c: Make the lock in the threadstorage + debugging code untracked to avoid a deadlock on thread + destruction. (closes issue #11207) Reported by: ys Patches: + threadstorage.c.diff uploaded by ys (license 281) Also fixes an + open bug report: (closes issue #11446) + + * main/utils.c: When DEBUG_THREADS is enabled, we only have the + details about who is holding a lock that we are waiting on for a + mutex, not rwlocks. This should fix the problem where people have + reported "core show locks" crashing sometimes. + + * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were + reported as happening on Mac systems. It turns out that the + problem was the Mac version of the ast_atomic_fetchadd_int() + function. The Mac atomic add function returns the _new_ value, + while this function is supposed to return the old value. So, the + crashes happened on unreferencing objects. If the reference count + was decreased to 1, ao2_ref() thought that it had been decreased + to zero, and called the destructor. However, there was still an + outstanding reference around. (closes issue #11176) (closes issue + #11289) + + * include/asterisk/file.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/compiler.h: Modify file.h to maintain API + compatibility with earlier versions. If a recent compiler is + being used, then a warning will show up for any modules still + using the old name "private" instead of "_private". (patch + suggested by paravoid) + + * main/pbx.c: Make some changes to some additions I made recently + for doing channel autoservice when looking up extensions. This + code was added to handle the case where a dialplan switch was in + use that could block for a long time. However, the way that I + added it, it did this for all extension lookups. However, lookups + in the in-memory tree of extensions should _not_ take long enough + to matter. So, move the autoservice stuff to be only around + executing a switch. + +2007-12-04 17:28 +0000 [r90876] Jason Parker + + * main/channel.c: If we fail to create a channel after allocating a + timing fd, we need to make sure to close it. Issue 11454, patch + by eliel. + +2007-12-04 05:29 +0000 [r90798] Joshua Colp + + * apps/app_dial.c: Fix build issue on the build cluster. + +2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher + + * include/asterisk/compat.h: Solaris requires the inclusion of + sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: + snuffy,tilghman (Closes issue #11430) + + * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a + NULL deref could result Reported by: xrg Patch by: tilghman + (Closes issue #11387) + +2007-12-03 23:12 +0000 [r90735] Mark Michelson + + * apps/app_dial.c, main/channel.c, main/global_datastores.c + (added), channels/chan_local.c, main/Makefile, + include/asterisk/channel.h, include/asterisk/global_datastores.h + (added), apps/app_queue.c: A big one... This is the merge of the + forward-loop branch. The main change here is that call-forwards + can no longer loop. This is accomplished by creating a datastore + on the calling channel which has a linked list of all devices + dialed. If a forward happens, then the local channel which is + created inherits the datastore. If, through this progression of + forwards and datastore inheritance, a device is attempted to be + dialed a second time, it will simply be skipped and a warning + message will be printed to the CLI. After the dialing has been + completed, the datastore is detached from the channel and + destroyed. This change also introduces some side effects to the + code which I shall enumerate here: 1. Datastore inheritance has + been backported from trunk into 1.4 2. A large chunk of code has + been removed from app_dial. This chunk is the section of code + which handles the call forward case after the channel has been + requested but before it has been called. This was removed because + call-forwarding still works fine without it, it makes the code + less error-prone should it need changing, and it made this set of + changes much less painful to just have the forwarding handled in + one place in each module. 3. Two new files, global_datastores.h + and .c have been added. These are necessary since the datastore + which is attached to the channel may be created and attached in + either app_dial or app_queue, so they need a common place to find + the datastore info. This approach was taken in case similar + datastores are needed in the future, there will be a common place + to add them. + +2007-12-03 22:06 +0000 [r90696] Jason Parker + + * apps/app_meetme.c: Make sure we always close the conference fd if + we have an open one. Issue 11383, reported by markmhy, patch by + eliel. + +2007-12-03 20:59 +0000 [r90639] Mark Michelson + + * channels/chan_mgcp.c: Changing some bad logic when calculating + the interdigit timeout. (closes issue #11402, reported and + patched by eferro) + +2007-12-03 20:51 +0000 [r90607] Jason Parker + + * res/res_features.c: Fix crash in ParkAndAnnounce application. + Issue #11436, reported by lytledd, patch by eliel. + +2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp + + * main/rtp.c: Do not create a smoother for G723.1 frames, they need + to be left alone to their native 20/24 byte size. + + * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve + the indication currently playing on a channel when a masquerade + operation happens. (issue #BE-88) + +2007-12-03 18:20 +0000 [r90546] Jason Parker + + * channels/chan_iax2.c: Only log debug messages if debug is + enabled. Closes issue #11416, patch by casper. + +2007-12-02 18:18 +0000 [r90470] Russell Bryant + + * apps/app_queue.c: The other day when I went through making + changes as a result of the ao2_link() change, I added some code + to set pointers to NULL after they were unreferenced. This + pointed out that in this place, the object was unreferenced + before the code was done using it. So, move the unref down a + little bit. (crash reported by jmls on IRC) + +2007-12-02 09:34 +0000 [r90432] Tilghman Lesher + + * main/autoservice.c: Clarify the return value on autoservice. + Specifically, if you started autoservice and autoservice was + already on, it would erroneously return an error. Reported by: + adiemus Patch by: dimas (Closes issue #11433) + +2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant + + * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h, + apps/app_queue.c, channels/chan_iax2.c: Change the behavior of + ao2_link(). Previously, in inherited a reference. Now, it + automatically increases the reference count to reflect the + reference that is now held by the container. This was done to be + more consistent with ao2_unlink(), which automatically releases + the reference held by the container. It also makes it so it is no + longer possible for a pointer to be invalid after ao2_link() + returns. + + * include/asterisk/astobj2.h: Add some notes on the behavior of + ao2_unlink() after a discussion with Tilghman + +2007-11-30 14:43 +0000 [r90269] Joshua Colp + + * channels/chan_sip.c: Fix locking issues under one legged replaces + scenarios. (closes issue #11420) Reported by: irroot Patches: + chan_sip_oneleg.patch uploaded by irroot (license 52) + +2007-11-30 00:16 +0000 [r90231] Mark Michelson + + * channels/chan_mgcp.c: Clear the DTMF buffer if the call times + out. (closes issue #11418, reported and patched by eferro) + +2007-11-29 Russell Bryant + + * Asterisk 1.4.15 released. + +2007-11-29 19:48 +0000 [r90166] Tilghman Lesher + + * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure + we use thread-safe escaping (Fixes AST-2007-026) + +2007-11-29 19:38 +0000 [r90163] Mark Michelson + + * apps/app_queue.c: This patch handles the case where a queue + member with a negative penalty is added via the manager. If a + negative value is submitted for a member penalty, we set it to 0. + (closes issue #11411, reported and patched by Laureano) + +2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher + + * res/res_config_pgsql.c: Properly escape input buffers (Fixes + AST-2007-025) + + * formats/format_g726.c, include/asterisk/file.h, + formats/format_wav.c, formats/format_pcm.c, + formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c, + formats/format_h264.c, formats/format_wav_gsm.c: Use of "private" + as a field name in a header file messes with C++ projects + Reported by: chewbacca Patch by: casper (Closes issue #11401) + + * sounds/Makefile: Upgrade the core sounds release version + +2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant + + * funcs/func_callerid.c: fix some formatting i accidentally changed + + * funcs/func_callerid.c, main/channel.c, + include/asterisk/channel.h: This set of changes is to make some + callerID handling thread-safe. The ast_set_callerid() function + needed to lock the channel. Also, the handlers for the CALLERID() + dialplan function needed to lock the channel when reading or + writing callerid values directly on the channel structure. + + * include/asterisk/file.h, main/file.c: Merge a change from + team/russell/chan_refcount ... This makes ast_stopstream() + thread-safe. + +2007-11-28 22:59 +0000 [r90101] Joshua Colp + + * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405) + Reported by: eliel Patches: load_realtime.patch uploaded by eliel + (license 64) + +2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming + + * configs/users.conf.sample, main/manager.c: it is impossible to + set permissions for manager accounts created by users.conf + (reported internally, patched by me) + +2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson + + * main/pbx.c: Removing some seemingly pointless code. This sets a + channel variable for every priority executed in the dialplan if + you have debug set to anything non-zero. This seems pointless due + to the fact that these channel variables are not referenced + anywhere else in the code and their names are esoteric enough + that they would not be practical to reference in the dialplan. + Plus the fact that this behavior isn't documented anywhere means + that the change is not likely to cause any disruption. If + anything, this may actually cause a slight performance increase + if running with debug on. The motivating influence for this code + change is the eventwhencalled option for queues. If set to vars, + all channel variables will be output to the manager. These + unnecessary channel variables make the output a lot more + difficult to deal with. + + * apps/app_voicemail.c: Recording greetings when using IMAP storage + was causing zero-length files to be stored. Since greetings are + not retrieved from IMAP anyway, it is pointless to attempt + storing them there. (closes issue #11359, reported by spditner, + patched by me) + +2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant + + * main/pbx.c, include/asterisk/pbx.h: - update documentation for + some of the goto functions to note that they handle locking the + channel as needed - update ast_explicit_goto() to lock the + channel as needed + + * main/autoservice.c: Don't do frame processing if ast_read() + returned NULL. + + * apps/app_queue.c: Instead of depending on the return value of + ast_true(), explicitly set the eventwhencalled variable to 1. + + * main/pbx.c: Don't start/stop autoservice in + pbx_extension_helper() unless a channel exists + +2007-11-27 23:10 +0000 [r89837] Mark Michelson + + * apps/app_queue.c: Two changes with regards to the + 'eventwhencalled' option of queues.conf 1) Due to some signed vs. + unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' + did exactly the same thing. Thus the sign change of the ast_true + call. 2) The vars2manager function overwrote a \n for every + channel variable it parsed, resulting in bizarre output for the + channel variables. This patch remedies this. (related to issue + #11385, however I'm not sure if this will actually be enough to + close it) + +2007-11-27 21:45 +0000 [r89790] Russell Bryant + + * main/autoservice.c, main/pbx.c: Merge changes from + team/russell/autoservice_1.4 This set of changes fixes an issue + that was reported to me on IRC yesterday. The user, d1mas, was + using chan_zap for incoming calls and was having DTMF recognition + issues in some situations. Specifically, he noticed that the + problem occurred when using DISA or WaitExten. He also noticed + that when using Read, the problem did not occur. His system also + used DUNDi for dialplan lookups. So, he theorized that if the + DUNDi lookups blocked for some period of time, that audio from + the zap channel could get lost. If the audio got lost, then it + wouldn't be run through the DTMF detector, and digits could get + lost. He was correct, and the following set of changes fixes the + problem. However, the changes go a little bit further than what + was necessary to fix this exact problem. 1) I updated + pbx_extension_helper() to autoservice the associated channel to + handle cases where extension lookups may take a long time. This + would normally be a dialplan switch that does some lookup over + the network, such as the DUNDi or IAX2 switches. This ensures + that even while a DUNDi lookup is blocking, the channel will be + continuously serviced. 2) I made a change to the autoservice + code. This is actually something that has bothered me for a long + time. When a channel is in autoservice, _all_ frames get thrown + away. However, some frames really shouldn't be thrown away. The + most notable examples are signalling (CONTROL) frames, and DTMF. + So, this patch queues up important frames while a channel is in + autoservice. When autoservice is stopped on the channel, the + queued up frames get stuck back on the channel so that they can + get processed instead of thrown away. 3) I made another change to + the autoservice code to handle the case where autoservice is + started on channels recursively. Previously, you could call + ast_autoservice_start() multiple times on a channel, and it would + stop the first time ast_autoservice_stop() gets called. Now, it + will ensure that autoservice doesn't actually stop until the + final call to ast_autoservice_stop(). + +2007-11-27 20:22 +0000 [r89727] Mark Michelson + + * res/res_config_pgsql.c: Changing some calls from free() to + ast_free() since they were allocated with ast_calloc(). (closes + issue #11390, reported and patched by Laureano) + +2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming + + * main/app.c: on second thought... revert all the other changes + i've made in app options parsing leaving only one: if an empty + argument is supplied for an option, set that argument pointer to + point to an empty string rather than NULL, so that the + application can do normal checks on it without worrying about it + being NULL + + * main/app.c: generate a warning when an application option that + requires an argument is ignored due to lack of an argument + +2007-11-27 16:12 +0000 [r89634] Russell Bryant + + * configs/voicemail.conf.sample: Add a note to the sample voicemail + config noting that when using IMAP storage, only the first format + specified will be attached to the message. + +2007-11-27 15:38 +0000 [r89631] Tilghman Lesher + + * funcs/func_env.c: Default result of STAT should be "0" not "". + Reported via the -users mailing list, fixed by me. + +2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we + get a codec offer using a well-known payload type, but using it + for another codec that we don't know, Asterisk did not remove + that codec from the list. With this patch, we remove the codec + from audio and video rtp objects and deny it ever existed. Thanks + to lasse for testing. (closes issue #11376) Reported by: lasse + Patches: bug11376.txt uploaded by oej (license 306) Tested by: + lasse + + * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue + #11304) Reported by: pj + +2007-11-27 06:24 +0000 [r89622] Steve Murphy + + * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample, + include/asterisk/cdr.h: closes issue #11379; OK, this is an + attempt to make both sides happy. To the cdr.conf file, I added + the option 'unanswered', which defaults to 'no'. In this mode, + you will see a cdr for a call, whether it was answered or not. + The disposition will be NO ANSWER or ANSWERED, as appropriate. + The src is as you'd expect, the destination channel will be one + of the channels from the Dial() call, usually the last in the + list if more than one chan was specified. With unanswered set to + 'yes', you will still see this cdr entry in both cases. But in + the case where the dial timed out, you will also see a cdr for + each line attempted, marked NO ANSWER, with no destination + channel name. The new option defaults to 'no', so you don't see + the pesky extra cdr's by default, and you will not see the + irritating 'not posted' messages. + +2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson + + * apps/app_playback.c: After issuing a "say load new", if a caller + hangs up during the middle of playback of a number, app_playback + will continue to try to play the remaining files. With this + change, no more files will be played back upon hangup. (closes + issue #11345, reported and patched by IgorG) + + * apps/app_playback.c: After issuing a "say load new" tons of + warning messages are printed out to the CLI every time do_say in + app_playback is called. Removing these warnings + +2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp + + * main/dial.c: Fix issues with async dialing with an application + executing. The application has to be terminated and control + returned to the thread before hanging things up. (issue #BE-252) + + * res/res_features.c: Add module counting removal for error + conditions. (closes issue #11333) Reported by: Laureano Patches: + res_features_v2.c.patch uploaded by Laureano (license 265) + +2007-11-26 17:41 +0000 [r89594] Russell Bryant + + * main/pbx.c: Add channel locking to a function that needed to be + doing it. This is just a little something I noticed while working + on a completely unrelated issue. + +2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp + + * pbx/pbx_config.c: Use ast_free to free memory, or else we shall + implode if MALLOC_DEBUG is enabled. (closes issue #11347) + Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys + (license 281) + + * apps/app_mixmonitor.c: Close the audio file before sending it to + the post processing application. (closes issue #11357) Reported + by: reformed Patches: mixmonitor.patch uploaded by reformed + (license 330) + +2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming + + * main/app.c: when parsing application options that take arguments, + don't indicate that the option was supplied unless a + non-zero-length argument was found for it + +2007-11-26 15:48 +0000 [r89580] Mark Michelson + + * apps/app_voicemail.c: Revert vmu->email back to an empty string + if it was empty when imap_store_file was called. This prevents + sending a duplicate e-mail. (closes issue #11204, reported by + spditner, patched by me) + +2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp + + * main/channel.c: If channel allocation fails because the alert + pipe could not be created also free the scheduler context. + (closes issue #11355) Reported by: eliel Patches: + main.channel.c.patch uploaded by eliel (license 64) + + * apps/app_meetme.c: When unloading app_meetme destroy any auto + created contexts created by SLA. (closes issue #11367) Reported + by: eliel + +2007-11-25 17:17 +0000 [r89559] Tilghman Lesher + + * res/res_odbc.c, configs/res_odbc.conf.sample, + include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously + attempted to use the ESCAPE clause to set the escape delimiter to + a backslash. Unfortunately, this does not universally work on all + databases, since on databases which natively use the backslash as + a delimiter, the backslash itself needs to be delimited, but on + other databases that have no delimiter, backslashing the + backslash causes an error. So the only solution that I can come + up with is to create an option in res_odbc that explicitly + specifies whether or not backslash is a native delimiter. If it + is, we use it natively; if not, we use the ESCAPE clause to make + it one. Reported by: elguero Patch by: tilghman (Closes issue + #11364) + +2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher + + * res/res_adsi.c: Free some frames that would otherwise leak on + error. Reported by: Laureano Patch by: Laureano,tilghman (Closes + issue #11351) + + * apps/app_voicemail.c, main/app.c: Currently, zero-length + voicemail messages cause a hangup in VoicemailMain. This change + fixes the problem, with a multi-faceted approach. First, we do + our best to avoid these messages from being created in the first + place, and second, if that fails, we detect when the voicemail + message is zero-length and avoid exiting at that point. Reported + by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) + + * main/manager.c: Up until this point, the XML output of the + manager has been technically invalid, due to the repetition of + certain parameters in a single event. This caused various issues + for XML parsers, some of which refused to parse at all, given the + invalidity of the rendered XML. So this commit fixes the XML + output, ensuring that each entity parameter has a unique name, + thus ensuring valid XML. Reported by: msetim Patch by: tilghman + (Closes issue #10220) + + * res/res_config_odbc.c: Use ESCAPE clause for the first parameter, + not just 2nd-Nth parameters. Reported by: apsaras Patch by: + tilghman (Closes issue #11353) + +2007-11-22 17:29 +0000 [r89527] Russell Bryant + + * configs/agents.conf.sample: mvanbaak pointed out a spelling error + in this sample configuration file. While I was at it, I went + ahead and tweaked it a little bit more. + +2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson + + * apps/app_queue.c: Fix a small error I made in my previous commit + + * apps/app_queue.c: Changing an inaccurate debug message to be less + inaccurate. Under the circumstances, this message would always + report that there were 0 members available, even though that may + not be true. + +2007-11-21 18:59 +0000 [r89491] Terry Wilson + + * res/res_features.c: If a channel gets masqueraded in the middle + of a park, don't play the announcement to the masqueraded + channel, and dial back to the original channel on timeout. + +2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming + + * include/asterisk/module.h: re-doxygen some comments + + * main/loader.c, include/asterisk/module.h, + build_tools/make_buildopts_h: bring back compile-option checking + when loading modules, only this time use a string-based storage + and comparison mechanism because it is easier to support on other + platforms + +2007-11-20 17:50 +0000 [r89457] Mark Michelson + + * main/pbx.c: According to comments in main/pbx.c, it is essential + that if we are going to lock the conlock as well as the hints + lock, it must be locked in that respective order. In order to + prevent a potential deadlock, we need to lock the conlock prior + to locking the hints lock in ast_hint_state_changed (see the call + stack example on issue #11323 for how this can happen). (closes + issue #11323, reported by eelcob, suggestion for patch by eelcob, + patch by me) + +2007-11-20 15:22 +0000 [r89450] Steve Murphy + + * doc/queues-with-callback-members.txt: closes issue #11324; break + statements missing in switch cases. + +2007-11-20 13:40 +0000 [r89445] Christian Richter + + * channels/chan_misdn.c: added RR patch from iroot #10908, thanks. + +2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp + + * res/res_features.c: Print out the correct filename + (features.conf) in the log message when parkpos options are + incorrect. (closes issue #11295) Reported by: Laureano Patches: + res_features.c.patch uploaded by Laureano (license 265) + + * doc/localchannel.txt: Clarify documentation a bit, include that a + frame has to pass through the core in order for the Local channel + optimization to happen. (closes issue #11246) Reported by: jon + +2007-11-16 Russell Bryant + + * Asterisk 1.4.14 released. + +2007-11-16 22:26 +0000 [r89339] Russell Bryant + + * main/loader.c, include/asterisk/module.h, + build_tools/make_buildopts_h: Temporarily revert revision 89325, + which added md5 magic for keeping track of what build options + were used. We agreed that we should remove this before making a + 1.4 release, and then we can put it back in. Then, we can take a + month or so to play around with it to get it how we want it. + +2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming + + * main/loader.c, include/asterisk/module.h, + build_tools/make_buildopts_h: To help combat problems where + people build external modules (asterisk-addons or others) and + then change the build options of the Asterisk build in a way that + makes the incompatible without warning, this commit introduces an + MD5 signature of the important build-time options and includes + that signature into modules when they are built. When the loader + loads one of these modules and notices the problem, it will emit + a warning to console and refuse to initialize the module, as + doing so could cause the system to be unstable or even crash. If + you upgrade to this version of Asterisk, you must rebuild *all* + of your modules that came from other sources before trying to run + this version. If you are using Digium's G.729 binary codec + module, you will need v33 or newer. + +2007-11-16 15:28 +0000 [r89323] Mark Michelson + + * apps/app_queue.c: Make realtime queues accessible from the + QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and + patched by atis, with small modifications from me) + +2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher + + * Makefile: Start Asterisk in Debian at a more reasonable time + (since zaptel is at level 20) + + * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found + by valgrind + + * channels/chan_iax2.c: Yet another memory corruption issue. + Reported by: atis Patch by: tilghman Fixes issue #10923 + +2007-11-15 17:19 +0000 [r89296] Russell Bryant + + * apps/app_meetme.c: Update the SLAStation application to account + for the case where the SLA thread has a call out to the station, + but the user has pressed a line button to answer the call instead + of picking up the handset. If they do, the phone sends out a new + INVITE. So, the SLAStation app must check to see if it is picking + up a ringing trunk, and ensure that the other stations stop + ringing. (reported internally, patched by me, tested by mogorman) + +2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson + + * main/manager.c: Undoing previous commit since I realize it was + wrong + + * main/manager.c: Adding a missing mutex unlock. (closes issue + 11256, reported and patched by ys) + +2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson + + * channels/chan_sip.c: Don't send re-invites during pending INVITE + transactions. Patch by one47 - thanks! Closes issue #9305 + + * channels/chan_sip.c: Improve support for multipart messages. Code + by gasparz, changes by me (mostly formatting). Thanks, gasparz! + Closes issue #10947 + +2007-11-14 23:23 +0000 [r89275] Tilghman Lesher + + * main/app.c: When a recording ends with '#', we are improperly + trimming an extra 200ms from the recording. Reported by: sim + Patch by: tilghman Closes issue #11247 + +2007-11-14 01:15 +0000 [r89260] Joshua Colp + + * main/srv.c: Return the proper value when the srv_callback + function executes properly. (closes issue #11240) Reported by: + jtodd + +2007-11-13 21:07 +0000 [r89248-89254] Jason Parker + + * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer + systems which require a third arg to open() when using O_CREAT. + Issue 11238, reported by puzzled. + + * res/res_features.c: Revert change from revision 67064. It is + documented behavior that if a parking extension already exists + while using PARKINGEXTEN, dialplan execution will continue. If + blind transferring to a Park with PARKINGEXTEN, you must keep + this in mind, and handle the failure yourself. Issue 11237, + reported by jon. + +2007-11-13 17:34 +0000 [r89246] Tilghman Lesher + + * channels/chan_sip.c: If we set a value for qualify, we should + actually pay attention to it, instead of overriding the value + +2007-11-13 16:02 +0000 [r89241] Mark Michelson + + * apps/app_mixmonitor.c: Reverting commit made in revision 89205 + since it is unnecessary. Thanks to Kevin for pointing this out + +2007-11-13 13:51 +0000 [r89239] Tilghman Lesher + + * main/utils.c: Debugging is running into the 16-lock limit. + Increase to avoid. (This define is only effective when debugging + is turned on, so there's no effect for most installations.) + +2007-11-13 00:56 +0000 [r89205] Mark Michelson + + * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If + only 1 argument is given, then the args.options and + args.post_process strings are uninitialized and could contain + garbage. This change handles this situation properly by only + using arguments that we have parsed. + +2007-11-12 20:46 +0000 [r89194] Jason Parker + + * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev + +2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher + + * main/config.c: If two config writes collide, file corruption + could result. Use a mkstemp() file, instead. Reported by: + paravoid Patch by: tilghman Closes issue #10781 + + * main/channel.c, channels/chan_sip.c: Fix two cases of memory + corruption caused by background threads. Reported by: atis Patch + by: tilghman Fixes issue #10923 + +2007-11-12 11:26 +0000 [r89169-89173] Christian Richter + + * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and + no number was dialed and overlapdial is set, we wait for the ISDN + timeout instead of starting our own timer. added a comment for + the misdn.conf.sample for the overlapdial config option. + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, + channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added + restart all interfaces Restart_Indicator, to automatically send a + RESTART after the L2 of a PTP Port comes up. Also fixed some + places where we have send a RELEASE without need for it. + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a + state/event issue with overlapdial=yes when no extension matched. + removed the general sending of a RELEASE_COMPLETE when we receive + a RELEASE, this is done by mISDNuser/mISDN. This makes it + possible to use asterisk-1.4 with mISDN trunk, but requires users + of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 + (when using the NT mode at all) + + * channels/misdn/isdn_lib.c: fixed the support for CW and therefore + for the reject_cause option. + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + aded ntkeepcalls option, to avoid droÃpping calls when the L2 + goes down on a PTP link. There are some pbx which do turn off the + L1 for a very short while and restart it immediately. normally + T310 should be started and after 10 seconds or so the calls + should be dropped, this is a simple fix wihtout this timer. + +2007-11-08 23:52 +0000 [r89125] Jason Parker + + * main/say.c: Properly say the seconds here.. Issue 11203, fix + described by vma. + +2007-11-08 21:00 +0000 [r89119] Mark Michelson + + * channels/chan_sip.c: Rework of the commit I made yesterday to use + the already built-in ast_uri_decode function as opposed to my + home-rolled one. Also added comments. Thanks to oej for pointing + me in the right direction + +2007-11-08 18:45 +0000 [r89115] Jason Parker + + * configs/res_odbc.conf.sample: Avoid warnings on load when using + sample configuration files. Issue 11195, patch by eliel. + +2007-11-08 16:47 +0000 [r89111] Mark Michelson + + * apps/app_voicemail.c: I made this same adjustment in trunk to fix + a bug, and it makes sense to do it in 1.4 as well. If an + imapfolder is specified in voicemail.conf, don't ever explicitly + connect to INBOX since it may not exist. + +2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming + + * main/srv.c: fix a glaring bug in the new SRV record handling that + would cause incorrect weight sorting + +2007-11-08 04:55 +0000 [r89103] Tilghman Lesher + + * doc/valgrind.txt: Typo + +2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp + + * channels/chan_sip.c: Do not add a sip: to the beginning of the To + URI unless needed. (closes issue #10756) Reported by: goestelecom + + * channels/chan_sip.c: Improve the devicestate logic for multiple + devices. If any are available then the extension is considered + available. (closes issue #10164) Reported by: nic_bellamy + Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic + (license 299) + + * channels/chan_sip.c: Add support for allowing one outgoing + transaction. This means if a response comes back out of order + chan_sip will still handle it. I dream of a chan_sip with real + transaction support. (closes issue #10946) Reported by: flefoll + (closes issue #10915) Reported by: ramonpeek (closes issue #9567) + Reported by: atca_pres + + * channels/chan_sip.c: If callerid is configured in sip.conf use + that for checking the presence of an extension in the dialplan. + (closes issue #11185) Reported by: spditner + +2007-11-07 23:39 +0000 [r89093] Tilghman Lesher + + * apps/app_queue.c: The member refcount must be incremented, to + avoid using it after deallocation. A huge thanks go to lvl- for + patiently providing the necessary valgrind output that was + necessary to finding this problem of memory corruption. Reported + by: lvl- Patch by: tilghman Closes issue #11174 + +2007-11-07 22:40 +0000 [r89090] Mark Michelson + + * channels/chan_sip.c: This patch makes it possible for SIP phones + to dial extensions defined with '#' characters in extensions.conf + AND maintain their escaped characters when forming URI's (closes + issue #10681, reported by cahen, patched by me, code review by + file) + +2007-11-07 21:40 +0000 [r89088] Steve Murphy + + * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to + 10578, I just ran 1.4 thru valgrind; some of the config leakage + I've already fixed, but it doesn't hurt to double check. I found + and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, + tho. + +2007-11-07 15:56 +0000 [r89085] Mark Michelson + + * main/manager.c: Fixing a segfault in the manager "core show + channels concise" command. (closes issue #11183, reported by arnd + and patched by ys) + +2007-11-07 04:07 +0000 [r89079] Tilghman Lesher + + * configs/extensions.ael.sample: Suppress AEL warnings on load. + Reported by: eliel Patch by: eliel Closes issue #11178 + +2007-11-06 20:18 +0000 [r89053] Russell Bryant + + * res/res_musiconhold.c: Fix init_classes() so that classes that + actually do have files loaded aren't treated as empty, and + immediately destroyed ... + +2007-11-06 19:09 +0000 [r89046] Jason Parker + + * codecs/codec_zap.c: Correctly set the total number of channels + from a zaptel transcoder board. SPD-49, patch by Matthew + Nicholson. + +2007-11-06 19:09 +0000 [r89045] Tilghman Lesher + + * include/asterisk/lock.h: We went to the trouble of creating a + method of tracking failed trylocks, then never turned it on + (oops). + +2007-11-06 18:53 +0000 [r89042] Olle Johansson + + * main/tdd.c: Bug fixes to tdd support in zaptel. + +2007-11-06 18:20 +0000 [r89037] Russell Bryant + + * res/res_musiconhold.c: If someone were to delete the files used + by an existing MOH class, and then issue a reload, further use of + that class could result in a crash due to dividing by zero. This + set of changes fixes up some places to prevent this from + happening. (closes issue #10948) Reported by: jcomellas Patches: + res_musiconhold_division_by_zero.patch uploaded by jcomellas + (license 282) Additional changes added by me. + +2007-11-06 17:52 +0000 [r89036] Steve Murphy + + * main/config.c: closes issue #8786 - where the [catname](!) and + [catname](othercat1,othercat2,...) notation gets dropped across a + ConfigUpdate (or any other thing that would cause a config file + to be written). While I was at it, I also cleaned up some of the + destroy routines to free up comments, which was not being done. + Made sure the new struct I introduced is also cleaned up properly + at destruction time. My code handles multiple template + inclusions. Many thanks to ssokol for his patch, which, while not + literally used in the final merge, served as a foundation for the + fix. + +2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp + + * channels/chan_sip.c: Make it so that if a peer is determined to + be unreachable using qualify their devicestate will report back + unavailable. (closes issue #11006) Reported by: pj + + * channels/chan_zap.c: Fix improbable but possible memory leaks in + chan_zap. (closes issue #11166) Reported by: eliel Patches: + chan_zap.c.patch uploaded by eliel (license 64) + +2007-11-06 13:50 +0000 [r88931] Russell Bryant + + * include/asterisk/lock.h: Remove some checks to see if locks are + initialized from the non-DEBUG_THREADS versions of the lock + routines. These are incorrect for a number of reasons: - It + breaks the build on mac. - If there is a problem with locks not + getting initialized, then the proper fix is to find that place + and fix the code so that it does get initialized. - If additional + debug code is needed to help find the problem areas, then this + type of things should _only_ be put in the DEBUG_THREADS + wrappers. + +2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming + + * include/asterisk/srv.h: update comment to match the state of the + code + +2007-11-05 23:29 +0000 [r88826] Mark Michelson + + * main/channel.c: Reworked deadlock avoidance in __ast_read. + Restored audio to callback agents. (closes issue #11071, reported + by callguy, patched by me, tested by callguy and Ted Brown) + +2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant + + * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related + to channel variables, I went looking around at the ways that + channel variables are handled. In general, they were not handled + in a thread-safe way. The channel _must_ be locked when reading + or writing from/to the channel variable list. What I have done to + improve this situation is to make pbx_builtin_setvar_helper() and + friends lock the channel when doing their thing. Asterisk API + calls almost all lock the channel for you as necessary, but this + family of functions did not. (closes issue #10923, reported by + atis) (closes issue #11159, reported by 850t) + + * channels/chan_sip.c: When traversing the list of channel + variables here in transmit_invite(), the asterisk channel must be + locked, as this data may change at any time. (I have seen + numerous reports of crashes related to the handling of channel + variables. There are a couple of issues on the bug tracker + related to it, but it has also been noted on IRC and mailing + lists. So, I am finding and fixing some places where channel + variables are handled improperly.) + + * channels/chan_sip.c: Fix up some indentation. + + * main/srv.c, include/asterisk/srv.h: Merge changes from + asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV + record support in Asterisk was broken. There was no guarantee on + what record Asterisk would choose to actually use. This set of + changes improves the situation by ensuring that Asterisk will + choose the highest priority record. + + * main/channel.c: Merge the last bit of changes from + asterisk/team/russell/readq-1.4 The issue here is that the + channel frame readq handling got broken when the code was + converted to use the linked list macros. It caused corruption of + the list head and tail pointers. So, I fixed up the usage of the + linked list macros and in passing, simplified the code. I also + documented what the code is doing, as it was a bit difficult to + figure out at first. This bug showed itself with crashes showing + messed up head/tail pointers for the readq. However, there are a + couple of crashes that aren't quite as obvious, but I think may + be related. So, if your bug gets closed by this commit, but you + still have a problem, please reopen or create a new bug report. + (closes issue #10936) (closes issue #10595) (closes issue #10368) + (closes issue #11084) (closes issue #10040) (closes issue #10840) + +2007-11-05 18:47 +0000 [r88671] Joshua Colp + + * channels/chan_sip.c: If a SIP channel is put on hold multiple + times do not keep incrementing the onHold value. (closes issue + #11085) Reported by: francesco_r Tested by: blitzrage (closes + issue #10474) Reported by: acennami + +2007-11-05 17:46 +0000 [r88624] Russell Bryant + + * main/channel.c: Fix up datastore handling in ast_do_masquerade(). + The code is intended to move any channel datastores from the old + channel to the new one. However, it did not use the linked list + macros properly to accomplish the task. The existing code would + only work if there was only a single datastore on the old + channel. + +2007-11-05 17:19 +0000 [r88585] Jason Parker + + * channels/chan_sip.c: Make sure we destroy the config structure on + configuration failure. Issue 11163, patch by eliel. + +2007-11-05 16:20 +0000 [r88539] Tilghman Lesher + + * res/res_odbc.c: Don't check used pooled connections for + connection status, as it will cause issues for prepared queries. + Reported by: Nick Gorham (via -dev list) Patch by: tilghman + +2007-11-04 22:38 +0000 [r88471] Luigi Rizzo + + * include/asterisk/stringfields.h, main/channel.c, + apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c: + Rename ast_string_field_free_pool to + ast_string_field_free_memory, and ast_string_field_free_all to + ast_string_field_reset_all to avoid misuse (due to too similar + names and an error in documentation). Fix two related memory + leaks in app_meetme. No need to merge to trunk, different fix + already applied there. Not applicable to 1.2 + +2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp + + * channels/chan_sip.c: Make subscribecontext behave as advertised. + It will now look for the presence of a hint in the given context + (be it subscribecontext or context). (closes issue #10702) + Reported by: slavon + + * channels/chan_sip.c: If an INFO request within a dialog is + received with a content length of 0 simply send back a 200 OK. It + is valid to do this and the remote side is probably using it to + make sure the signalling is still alive. (closes issue #5747) + Reported by: chandi Patches: infofix-81430-1.patch uploaded by + IgorG (license 20) + +2007-11-02 16:51 +0000 [r88283] Jason Parker + + * main/say.c: We need to make sure to specify a language to + ast_fileexists, otherwise it may fail for anything besides en + Issue 11147, fix discovered by both citats and myself + (independently), with input from Corydon76 + +2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher + + * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy + Patch by: ys Closes issue #11143 + + * doc/valgrind.txt (added): Add some notes on using valgrind + +2007-11-01 16:21 +0000 [r88078] Jason Parker + + * channels/chan_zap.c: Make sure we set the poll fds to NULL after + free()ing it. Part of issue 11017, patch by tzafrir. + +2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp + + * apps/app_meetme.c: Fix up commit for my Zap channel with spies in + Meetme fix. (thanks Tony Mountifield!) + + * apps/app_meetme.c: If a Zap channel contains a spy or a spy is + added take it out of the conference in kernel space and make it + go through Asterisk so the spy gets audio from both sides. + (closes issue #10060) Reported by: mparker + +2007-10-31 21:23 +0000 [r87906-87908] Jason Parker + + * res/res_jabber.c: Make sure we free some allocated memory before + returning. Issue 11131, patch by eliel. + + * channels/chan_gtalk.c: Don't try to allocate memory that we're + just going to re-allocate later anyways. Issue 11130, patch by + eliel. + +2007-10-31 18:03 +0000 [r87852] Tilghman Lesher + + * Makefile: Create samples for ALL of the available options in + asterisk.conf + +2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy + + * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan + save' cli command saves a file where the semicolon is not + escaped. Fixed this; User also wanted comments to be preserved + across dialplan save, but this is impossible at this point in + time, because comments are not stored in the dialplan. They are + 'compiled' out of extensions.conf. The only way to preserve those + comments is to use the config file reader/writer that the GUI + uses to allow online user edits. extensions.conf is first and + foremost, a config file, and is read in by the normal config-file + reading routines. Then, it is processed into a dialplan + (context/exten structs). + + * pbx/pbx_ael.c: Included some verbage in the check_includes func, + to inform the user that included contexts that have no match in + the AEL, might be OK, as AEL cannot check in the extensions.conf + or the in-memory contexts, as they may not be there at the time + of the check. + +2007-10-30 23:02 +0000 [r87739] Tilghman Lesher + + * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD + Reported by: ys Fixed by: ys Closes issue #11116 + +2007-10-30 21:19 +0000 [r87686] Russell Bryant + + * channels/chan_iax2.c: Merge the changes from + team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a + race condition related to the handling of POKEing peers. + Essentially, a reference to a peer is held by the scheduler when + there are pending callbacks, but the reference count didn't + reflect it. So, it was possible for a peer to hit a reference + count of zero and have its destructor begin to be called at the + same time that the scheduler thread ran a POKE related callback. + If that happened, a crash would likely occur. (closes issue + #11082, closes issue #11094) + +2007-10-30 20:29 +0000 [r87650] Jason Parker + + * channels/Makefile: Only try to clean out h323/ if the + h323/Makefile exists. + +2007-10-30 16:13 +0000 [r87571] Joshua Colp + + * res/res_features.c: Add two more checks before printing out a + warning message about bridging. If either channel has hungup of + course the bridge will have failed. (closes issue #10009) + Reported by: dimas + +2007-10-30 15:45 +0000 [r87567] Jason Parker + + * main/editline/np/vis.c: Fix build of editline on Solaris. Issue + 11113, patch by snuffy. + +2007-10-30 15:10 +0000 [r87534] Joshua Colp + + * apps/app_followme.c: Return 1.4 to a state where it builds. + Changing the arguments to a function and not changing where they + are used is bad, mmmk? + +2007-10-30 14:31 +0000 [r87514] BJ Weschke + + * apps/app_followme.c: Fix issue where the recorded name wasn't + getting removed correctly. (closes issue #11115) Reported by: + davevg Patches: followme-v3.diff + +2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming + + * codecs/gsm: missed one directory + + * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs, + codecs/lpc10, main/db1-ast, main/editline, main, + codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile, + codecs/lpc10/Makefile, utils, codecs, agi, + main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr: + clean up (and ignore) assembler and preprocessor intermediate + files if any are created during the build + + * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is + already there (used when debugging preprocessor issues) because + the compiler will whine about each compile command + +2007-10-29 21:06 +0000 [r87427] Mark Michelson + + * apps/app_voicemail.c: Removing a completely unnecessary quota + check from IMAP code. + +2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant + + * main/utils.c, include/asterisk/lock.h: Add some more details to + the output of "core show locks". When a thread is waiting for a + lock, this will now show the details about who currently has it + locked. (inspired by issue #11100) + + * main/astmm.c: Remove a lock that doesn't make any sense. The + regions lock needs to be held when traversing the list of + allocated chunks so that they can be printed out to the CLI. + (Thanks to eliel on #asterisk-dev for pointing this out!) + +2007-10-29 17:20 +0000 [r87342] Joshua Colp + + * channels/chan_sip.c: Fix issue where if both sides of the dialog + cancelled the dialog at the same time chan_sip could kepe + retransmitting a response for no reason. (closes issue #9566) + Reported by: atca_pres Patches: bug9566.patch uploaded by oej + +2007-10-29 17:13 +0000 [r87340] Jason Parker + + * funcs/func_realtime.c, funcs/func_cut.c: Allow some function + modules to compile under dev mode. Issue 11104, patch by andrew. + +2007-10-29 14:23 +0000 [r87294] Joshua Colp + + * main/utils.c: Fix issue with ast_unescape_semicolon going into an + endless loop. (closes issue #10550) Reported by: ramonpeek + Patches: unescape-85177-1.patch uploaded by IgorG (license 20) + +2007-10-28 13:46 +0000 [r87262] Tilghman Lesher + + * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c, + funcs/func_cut.c: Add autoservice to several more functions which + might delay in their responses. Also, make sure that func_odbc + functions have a channel on which to set variables. Reported by + russell Fixed by tilghman Closes issue #11099 + +2007-10-26 16:34 +0000 [r87168] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c, + pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c, + include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, + utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16, + pbx/ael/ael.flex: closes issue #11086 where a user complains that + references to following contexts report a problem; The problem + was REALLy that he was referring to empty contexts, which were + being ignored. Reporter stated that empty contexts should be OK. + I checked it out against extensions.conf, and sure enough, empty + contexts ARE ok. So, I removed the restriction from AEL. This, + though, highlighted a problem with multiple contexts of the same + name. This should be OK, also. So, I added the extend keyword to + AEL, and it can preceed the 'context' keyword (mixed with + 'abstract', if nec.). This will turn off the warnings in AEL if + the same context name is used 2 or more times. Also, I now call + ast_context_find_or_create for contexts now, instead of just + ast_context_create; I did this because pbx_config does this. The + 'extend' keyword thus becomes a statement of intent. AEL can now + duplicate the behavior of pbx_config, + +2007-10-26 13:54 +0000 [r87120] Tilghman Lesher + + * funcs/func_curl.c: The addition of autoservice to func_curl + additionally made func_curl dependent on the existence of a + channel, with no real reason. This should make func_curl once + again work without a channel. Reported by jmls. Fixed by + tilghman. Closes issue #11090 + +2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming + + * main/channel.c, include/asterisk/linkedlists.h: appending one + list to another should leave the first list empty, and not + require the user to do that + +2007-10-25 22:53 +0000 [r87067] Tilghman Lesher + + * funcs/func_cut.c: Backport alternate encoding of newline + delimiters from trunk to 1.4, as approved by Russell Reported by + blitzrage Closes issue #10903 + +2007-10-24 20:56 +0000 [r86982] Jason Parker + + * channels/chan_zap.c: Correctly respect hidecalleridname + configuration option. Simplify code slightly in the process. + Issue 11079, reported by ddv2005 + +2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy + + * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable + to specify app:spec in hint arguments + + * funcs/func_logic.c: closes issue #11052 -- where nothing after + the ? will allow un-initialized variable values to corrupt and + crash asterisk on 64-bit platforms + + * main/Makefile: this update to Makefile corrects how ast_expr2f.c + should be generated + + * main/ast_expr2f.c: This should get rid of a really, really + irritating warning generated by some 64-bit platforms from libc, + where free(0) is frowned upon + +2007-10-22 21:36 +0000 [r86836] Russell Bryant + + * include/asterisk/lock.h: If lock tracking is not enabled, then we + can not attempt to log any mutex failures. If so, we could end up + in infinite recursion. The only lock that is affected by this is + a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes + issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys + (license 281) + +2007-10-22 17:38 +0000 [r86787] Tilghman Lesher + + * main/astmm.c: Minor FreeBSD build fix + +2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp + + * channels/chan_sip.c: After reading online I have confirmed that + Record-Route headers should be copied to 1xx responses as well. + (closes issue #10113) Reported by: makoto + + * apps/app_controlplayback.c: Make sure res is a positive value + before performing the check to determine whether the user stopped + it or not. (closes issue #11023) Reported by: cfc + +2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant + + * main/channel.c: Don't leak a frame in the case that an END frame + is received and the time since the BEGIN is less than that of the + defined minimum DTMF duration. (closes issue #11051) Reported by: + casper Patches: channel.c.86664.diff uploaded by casper (license + 55) + + * include/asterisk/lock.h: Update the static mutex initializer to + include the initialization of the internal mutex used to protect + the lock debugging data. (closes issue #11044, patch suggested by + Ivan) + +2007-10-22 14:48 +0000 [r86694] Mark Michelson + + * apps/app_voicemail.c: Account for the fact that sometimes headers + may be terminated with \r\n instead of just \n (closes issue + #11043, reported by yehavi) + +2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp + + * main/channel.c: Move log message to before the frame it + references is freed. (closes issue #11050) Reported by: slavon + Patches: channel.c.86662.diff uploaded by casper (license 55) + + * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes + issue #11041) Reported by: jsmith Patches: + asterisk-dundicomplete.diff.txt uploaded by jamesgolovich + (license 176) + + * main/loader.c: Fixes for building under OpenSolaris. (closes + issue #11047) Reported by: snuffy Patches: 11047-fixes.diff + uploaded by snuffy (license 35) + +2007-10-22 09:21 +0000 [r86598] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send + DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan + does not match after an overlap call. Also added out_cause=1 + +2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp + + * main/app.c: When returning a DTMF digit from + ast_control_streamfile cast it as a char so that 0 does not + overlap with the success return code. (closes issue #11023) + Reported by: cfc + + * channels/chan_sip.c: Fix two issues with domains and transfers. + If a port was given in the hostname it was treated as part of the + hostname. If domains were configured but external domains were + not enabled all transfers would be considered remote. (closes + issue #11027) Reported by: ramonpeek Patches: 11027-1.diff + uploaded by ramonpeek (license 266) + + * channels/chan_sip.c: Set port number in received as information + for registrations as well. (closes issue #11028) Reported by: + brad-x + +2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development + + * apps/app_osplookup.c: Fixed OSP module did not report + source/devinfo IP in correct format. + +2007-10-18 22:01 +0000 [r86405-86406] Jason Parker + + * Makefile: Correct documentation. I removed the wrong line.. + + * Makefile: Add documentation for options in asterisk.conf Issue + 11029, patch by eserra + +2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant + + * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous + commit. + + * configs/iax.conf.sample, channels/chan_iax2.c: Add support for + setting the maximum trunk size for IAX2 trunking + + * main/channel.c, include/asterisk/channel.h: The channel needs to + stay locked while running timer callbacks, as they access and + modify channel data that may change elsewhere. I went through + every timer callback in the source tree to make sure that none of + them did any additional locking that could introduce deadlocks, + and all is well. (closes issue #10765) Reported by: Ivan Patches: + ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license + 229) + +2007-10-18 17:38 +0000 [r86328] Mark Michelson + + * apps/app_queue.c: If a non-existent file is specified to be + played either as a periodic announcement or as a hold/position + announcement, the caller would be kicked out of the queue. No + longer does this happen. + +2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant + + * codecs/codec_zap.c: Execute the RELEASE operation on transcoder + channels in the destroy callback. (patch from jsloan) + + * main/utils.c: Revert a change that I made for issue #10979 which, + as has been pointed out to me in issue #11018, doesn't really + make sense. There is no reason to have the base64 decode function + force a '\0' terminated buffer, when the result is almost always + binary, anyway. In fact, this caused some breakage, as some code + in res_crypto passed in a buffer exactly the right size to get + its binary result, which got stomped on by this patch. (closes + issue #11018, reported by dimas) + +2007-10-17 21:39 +0000 [r86202] Mark Michelson + + * apps/app_queue.c: Changing the strategy field of the call_queue + struct to be signed instead of unsigned, since the code attempts + to set the strategy to -1 if you specify a bogus strategy. While + this isn't a huge issue in 1.4, it could be a problem for someone + who, say, tries to use the roundrobin strategy in trunk (despite + all the deprecation warnings in 1.4). + +2007-10-17 17:57 +0000 [r86149] Russell Bryant + + * channels/chan_sip.c: If Asterisk is in the middle of shutting + down, respond to OPTIONS with 503 Unavailable. (closes issue + #10994) Reported by: eserra Patches: sip-options-503.patch + uploaded by eserra (license 45) + +2007-10-17 16:58 +0000 [r86117] Joshua Colp + + * channels/chan_sip.c: Whoops, forgot to remove the original + sip_scheddestroy. (closes issue #11010) Reported by: vadim + +2007-10-17 15:23 +0000 [r86066] Tilghman Lesher + + * main/asterisk.c: When runuser/rungroup is specified, a remote + console could only be attained by root (Closes issue #9999) + +2007-10-17 15:06 +0000 [r86063] Joshua Colp + + * channels/chan_sip.c: Don't schedule dialog destruction if a + MESSAGE is received using an existing dialog. (closes issue + #11010) Reported by: vadim + +2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson + + * configs/queues.conf.sample: Since monitor-join is deprecated now, + remove the example from the sample queues.conf file + + * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of + the monitor-join queue option + + * apps/app_queue.c: Adding deprecated warning to monitor-join + option, since the plan is to no longer support this in favor of + monitor-type = mixmonitor (related to issue #10885) + +2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant + + * include/asterisk/lock.h: really picky formatting tweak ... + + * include/asterisk/lock.h: Some locking errors exposed the fact + that the lock debugging code itself was not thread safe. How + ironic! Anyway, these changes ensure that the code that is + accessing the lock debugging data is thread-safe. Many thanks to + Ivan for finding and fixing the core issue here, and also thanks + to those that tested the patch and provided test results. (closes + issue #10571) (closes issue #10886) (closes issue #10875) (might + close some others, as well ...) Patches: (from issue #10571) + ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license + 229) - a few small changes by me + +2007-10-16 21:14 +0000 [r85958] Mark Michelson + + * apps/app_queue.c: Trying to remove a non-dynamic queue member via + dynamic means can lead to some interesting (read nasty) + situations. This patch clears up the issue by making only dynamic + queue members removable via dynamic methods. + +2007-10-16 19:41 +0000 [r85921] Tilghman Lesher + + * main/stdtime/localtime.c: Also set up gmtoff (this is used in the + %z gnu extension to strftime) Reported and fixed by jcmoore + Closes issue #11002 + +2007-10-16 19:10 +0000 [r85896] Russell Bryant + + * apps/app_voicemail.c: Remove a pointless lock. + +2007-10-16 15:21 +0000 [r85852] Mark Michelson + + * apps/app_queue.c: Fixing a double free which happens in the + statechange thread. (closes issue #10987, reported by andrew) + +2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp + + * apps/app_hasnewvoicemail.c: Check to make sure a value has been + given to the VMCOUNT dialplan function. (closes issue #10996) + Reported by: marsosa + + * main/threadstorage.c: Fix memory allocation issue in + threadstorage. (closes issue #10995) Reported by: snuffy Patches: + new-patch.diff uploaded by snuffy (license 35) + +2007-10-16 10:46 +0000 [r85800] Philippe Sultan + + * channels/chan_gtalk.c: Fix the output for this channel help CLI + command + +2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant + + * apps/app_queue.c: Ensure that no pending state changes are leaked + when the device state change thread gets stopped on module + unload. + + * apps/app_queue.c: Previously, app_queue created a thread to + handle every single device state change. I changed this a while + ago in trunk for performance reasons. However, bug 8407 points + out that it is actually a race condition, causing device state + changes to get processed in random order. So, I backported my + changes from trunk to 1.4. (closes issue #8407, patch provided by + tim_ringenbach, committed patch by me) + +2007-10-15 20:29 +0000 [r85687] Tilghman Lesher + + * apps/app_stack.c: Don't execute a gosub if the arguments is + zero-len (not just NULL) Reported by davevg Fixed by me Closes + issue #10985 + +2007-10-15 20:21 +0000 [r85686] Russell Bryant + + * main/say.c: Add a small fix for the tw version of saying dates. + (closes issue #7827) Reported by: sharkey Patches: say.nits.patch + uploaded by sharkey (license 172) + +2007-10-15 20:15 +0000 [r85684] Jason Parker + + * Makefile: Properly use DESTDIR in 'config' target. Do not try to + run chkconfig or similar if using DESTDIR. Issue 10938, patch by + cabal95. + +2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant + + * main/utils.c: Be pedantic about handling memory allocation + failure. + + * main/utils.c: The loop in the handler for the "core show locks" + could potentially block for some amount of time. Be a little bit + more careful and prepare all of the output in an intermediary + buffer while holding a global resource. Then, after releasing it, + send the output to ast_cli(). + + * channels/chan_sip.c: Make the default for the srvlookup option to + be yes. It doesn't really make sense for it to default to off. + The default configuration file has it on, and proper RFC + behavior, as indicated by a comment in the code, is for it to be + on. So, let's have it on by default to make lives easier. (closes + issue #10954, suggested by jtodd) + +2007-10-15 16:39 +0000 [r85571] Joshua Colp + + * configs/features.conf.sample: Document that DTMF based features + only work when two channels are bridged together. (closes issue + #10773) Reported by: pbayley + +2007-10-15 16:34 +0000 [r85561] Russell Bryant + + * include/asterisk/strings.h: Make a few changes so that characters + in the upper half of the ISO-8859-1 character set don't get + stripped when reading configuration. (closes issue #10982, + dandre) + +2007-10-15 16:22 +0000 [r85559] Joshua Colp + + * main/rtp.c: Bring both DTMF begin and end frames up through to + the core for DTMF feature handling. (closes issue #10826) + Reported by: dimas + +2007-10-15 15:40 +0000 [r85556] Russell Bryant + + * pbx/pbx_dundi.c: Ensure the buffer passed to + ast_canmatch_extension() is properly initialized so that it is + null terminated. (issue #10977) Reported by: dimas Patches: + pbxdundi.patch uploaded by dimas (license 88) - small mods by me + +2007-10-15 14:55 +0000 [r85552] Joshua Colp + + * main/rtp.c: If Monitor or a spy was added to a P2P or native + bridged channel bring the channel back to the generic bridging + core so the monitor or spy operations work. (closes issue #10943) + Reported by: julianjm + +2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant + + * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled. + (closes issue #10980) Reported by: casper Patches: + db.c.84633.diff uploaded by casper (license 55) + + * main/asterisk.c: Make sure remote consoles unmute themselves + again after reconnecting. (closes issue #10847) Reported by: atis + Patches: console_unmute_on_reconnect.patch uploaded by atis + (license 242) + + * main/utils.c: Make sure that the base64 decoder returns a + terminated string. (closes issue #10979) Reported by: ys Patches: + util.c.diff uploaded by ys (license 281) - small mods by me + + * pbx/pbx_config.c: Don't create the context for users in + users.conf until we know at least one user exists. (closes issue + #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by + dimas (license 88) + +2007-10-13 15:26 +0000 [r85536] Tilghman Lesher + + * configs/extensions.ael.sample: Remove deprecated syntax from + sample ael file Reported and patched by: dimas Closes issue + #10967 + +2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant + + * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an + issue with console verbosity when running asterisk -rx to execute + a command and retrieve its output. The issue was that there was + no way for the main Asterisk process to know that the remote + console was connecting in the -rx mode. The way that James has + fixed this is to have all remote consoles muted by default. Then, + regular remote consoles automatically execute a CLI command to + unmute themselves when they first start up. (closes issue #10847) + Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded + by jamesgolovich (license 176) + + * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly + handle the case where read() may return the text for more than + one CLI command at once for a remote console. (closes issue + #10888) Reported by: jamesgolovich Patches: + asterisk-climultiple.diff.txt uploaded by jamesgolovich (license + 176) + +2007-10-12 18:30 +0000 [r85523] Tilghman Lesher + + * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address + +2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant + + * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not + SDMI. (closes issue #10959) + + * pbx/pbx_realtime.c: Fix the potential use of an uninitialized + buffer in a log message. (closes issue #10958) Reported by: dimas + Patches: realtime.patch uploaded by dimas (license 88) + +2007-10-11 15:26 +0000 [r85397] Joshua Colp + + * channels/chan_sip.c: When creating a new packet don't try to stop + retransmission of it. It was just allocated/created so it's + impossible for it to have already been scheduled. (closes issue + #10945) Reported by: flefoll Patches: + chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll + (license 244) + +2007-10-11 04:35 +0000 [r85356] Tilghman Lesher + + * main/pbx.c: A dollar sign by itself, not indicating a start of a + variable or expression prematurely ends substitution (closes + issue #10939) + +2007-10-10 Russell Bryant + + * Asterisk 1.4.13 released. + +2007-10-10 15:56 +0000 [r85316] Russell Bryant + + * include/asterisk/file.h: I introduced a new member to the + ast_filestream struct in 1.4.12, but put it in the middle of the + struct, instead of at the end. One of the Debian folks, paravoid, + pointed out that this breaks binary compatability with modules + compiled against older headers. So, I'm moving the new member to + the end of the struct to resolve the situation. + +2007-10-10 15:51 +0000 [r85315] Mark Michelson + + * main/utils.c: The thread ID should be unsigned. + +2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp + + * channels/chan_sip.c: If devicestate is passed a port number strip + it out. (closes issue #10930) Reported by: ibc + + * channels/chan_sip.c: Add support for handling a 182 Queued + response. (closes issue #10924) Reported by: ramonpeek Patches: + queued-182.diff uploaded by ramonpeek (license 266) + +2007-10-10 14:26 +0000 [r85276] Mark Michelson + + * apps/app_voicemail.c: A bunch of changes from sprintf to + snprintf. See security advisory AST-2002-022 + +2007-10-10 14:14 +0000 [r85242] Joshua Colp + + * apps/app_voicemail.c: Close voicemail message description file if + duration did not meet the minimum, or else we will eventually run + out of file descriptors. (closes issue #10918) Reported by: + brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license + 279) + +2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming + + * include/asterisk/frame.h: use a macro instead of an inline + function, so that backtraces will report the caller of + ast_frame_free() properly + +2007-10-09 21:55 +0000 [r85158] Tilghman Lesher + + * main/channel.c, main/utils.c, include/asterisk/lock.h: This + commit fixes the following issues: - Deadlock in ast_write (issue + #10406) - Deadlock in ast_read (issue #10406) - Possible mutex + initialization error in lock.h (issue #10571) + +2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp + + * channels/chan_sip.c: Don't perform a reinvite if a transfer is in + progress. (issue #10915) Reported by: ramonpeek + + * main/rtp.c: Only update codec information if the channel has a + technology private structure. (issue #10915) Reported by: + ramonpeek + + * main/rtp.c: Update codec information as well as address when + doing hold reinvites. (issue #10868) Reported by: mavince + + * main/channel.c: Don't keep trying to native bridge if either of + the channels are involved in a masquerade operation to be done. + (closes issue #10696) Reported by: tbelder + +2007-10-08 03:28 +0000 [r84957] Russell Bryant + + * Makefile.rules: Enable file dependency tracking for _all_ builds, + and not just for builds with dev-mode enabled. I have seen enough + problems caused by this that I don't think it's worth keeping. I + want to continue to encourage anybody that is interested to + continue to run Asterisk from svn. Furthermore, I do not want + their systems to break when we change a structure definition in a + header file. :) + +2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan + + * res/res_jabber.c: Presence packets from a client who's connected + with our Jabber ID are valid, therefore, those clients must be + considered as buddies. The resource string helps us make the + distinction between clients. Closes issue #10707, reported by + yusufmotiwala. + + * res/res_jabber.c: Prevent Asterisk from crashing when receiving a + presence packet without resource from a buddy that is known to + have a resource list. Revert a change I previously made, where + Asterisk could point to a freed memory location. + +2007-10-05 19:42 +0000 [r84851] Tilghman Lesher + + * main/db.c: Log exactly why we can't open the database, if we fail + (closes issue #10887) + +2007-10-05 18:55 +0000 [r84818] Joshua Colp + + * main/rtp.c: Update the remembered RTP peer information when + putting an endpoint on hold or taking it off hold so that the RTP + stack does not initiate a needless reinvite. (closes issue + #10868) Reported by: mavince + +2007-10-05 16:44 +0000 [r84783] Russell Bryant + + * channels/chan_zap.c: Do deadlock avoidance in a couple more + places. You can't lock two channels at the same time without + doing extra work to make sure it succeeds. (closes issue #10895, + patch by me) + +2007-10-05 Russell Bryant + + * Asterisk 1.4.12.1 released. (This is mainly to include the + app_queue fix for a memory leak on reload, but includes a couple + of other bug fixes, as well.) + +2007-10-05 01:39 +0000 [r84742] Russell Bryant + + * main/manager.c: Fix a copy/paste error in the description of + UpdateConfig that was pointed out by JerJer on #asterisk-dev + +2007-10-04 21:57 +0000 [r84692] Mark Michelson + + * apps/app_queue.c: Don't allocate space for queue members unless + it's needed. You end up deleting dynamic members on a reload. Not + good. closes issue (#10879, reported by dazza76, patched by me) + +2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming + + * channels/chan_zap.c: callers of sig2str already add the word + 'signalling' in the appropriate place, so don't duplicate it + +2007-10-04 14:51 +0000 [r84637] Joshua Colp + + * apps/app_queue.c: Create a duplicate of the channel's member name + as the tab completion stuff will free it. (closes issue #10884) + Reported by: adamg + +2007-10-03 22:59 +0000 [r84581] Tilghman Lesher + + * main/rtp.c: When an RFC 2833 event is sent that we don't + recognize, ignore it, don't queue a NULL digit (closes issue + #10877) + +2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy + + * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call + in a switch expr doesn't execute correctly, because the commas in + the function args are not converted to vertbars before the func + is called. I modified just the switch code to convert the commas + to vertbars if there, but if more of these sort of probs are + found, I may have to resort to something a little more + fundamental. We'll see, I guess. + + * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, + pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a + null input to a switch statement results in a hangup; since + switch is implemented with extensions, and the default case is + implemented with a '.', and the '.' matches 1 or more remaining + characters, the case where 0 characters exist isn't matched, and + the extension isn't matched, and the goto fails, and a hangup + occurs. Now, when a default case is generated, it also generates + a single fixed extension that will match a null input. That + extension just does a goto to the default extension for that + switch. I played with an alternate solution, where I just tack an + extra char onto all the patterns and the goto, but not the + default case's pattern. Then even a null input will still have at + least one char in it. But it made me nervous, having that extra + char in , even if that's a pretty secret and low-level issue. + +2007-10-02 Russell Bryant + + * Asterisk 1.4.12 released. + +2007-10-02 20:06 +0000 [r84474] Russell Bryant + + * Makefile, build_tools/prep_tarball: * Don't build the + menuselect-tree for the tarball, as it requires running the + configure script first * Change the Makefile to note that + menuselect-tree depends on the configure script. + +2007-10-02 19:01 +0000 [r84410-84437] Jason Parker + + * res/res_features.c: Fix some odd formatting I missed.. + + * res/res_features.c: Finish up on transferee channel before return + on failure. Issue 10821, patch by Ivan + +2007-10-02 14:12 +0000 [r84370] Russell Bryant + + * channels/chan_sip.c: Use snprintf instead of sprintf in one + place. There is no vulnerability here due to various buffer sizes + around the code, but I still didn't like seeing a non + length-limited copy of data coming off of the wire into a stack + buffer, as this would be a problem in the future if buffer sizes + elsewhere got changed or size limitations removed ... + +2007-10-02 09:48 +0000 [r84345] Christian Richter + + * channels/chan_misdn.c: terminate USERUSER String with 0 + +2007-10-01 21:52 +0000 [r84291] Jason Parker + + * Makefile, Makefile.rules, channels/Makefile: Add dist-clean + support for subdirs. Change h323 to only remove the Makefile on a + dist-clean, rather than a clean. This fixes a bug I found with + trying to run make after a make clean + +2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard + + * main/channel.c, main/manager.c, channels/chan_agent.c: moved + get_base_channel() code from action_redirect to + ast_channel_masquerade() for issue 7706 and BE-160 + +2007-10-01 21:18 +0000 [r84273] Steve Murphy + + * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy... + +2007-10-01 21:07 +0000 [r84271] Russell Bryant + + * main/utils.c, include/asterisk/lock.h: Fulfull a feature request + from Qwell on the "core show locks" output. It will now note the + lock type for each lock that a thread holds. (mutex, rdlock, or + wrlock) + +2007-10-01 20:27 +0000 [r84239] Steve Murphy + + * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue + #10777 -- by returning a null for the parse tree when there's + really nothing there, and making sure we don't try to do checking + on a null tree. + +2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant + + * res/res_agi.c: Add another sanity check in the AGI read loop. We + really don't care about EAGAIN unless we didn't read an entire + line. If there is a newline at the end if the read buffer, break, + because we got the whole thing. (reported and patched by bmd) + + * include/asterisk/lock.h: Show rwlocks in the "core show locks" + output. Before, it only showed mutexes. + + * channels/Makefile: Remove another file in "make clean". (closes + issue #10814, paravoid) + + * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit. + +2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp + + * configs/usbradio.conf.sample (removed): Remove chan_usbradio + config file from tree, it is not present in here. (closes issue + #10839) Reported by: casper + + * res/res_musiconhold.c: Fix randomness. save_pos was being set to + 0 initially instead of -1, causing it to jump to position 0 when + moh started. (closes issue #10859) Reported by: jamesgolovich + Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich + (license 176) + + * apps/app_dial.c: Only attempt early bridging if the options given + to Dial() permit it. (closes issue #10861) Reported by: peekyb + +2007-09-30 20:02 +0000 [r84146] Russell Bryant + + * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++ + modules. The load and reload parameters were in the wrong place. + (closes issue #10846, alebm) + +2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy + + * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), + pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), + pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 + (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), + pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), + pbx/ael/ael-test/ref.ael-test4, + pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), + pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), + pbx/ael/ael-test/ael-ntest22/qq.ael (added), + pbx/ael/ael-test/ael-ntest22/t1 (added), + pbx/ael/ael-test/ael-ntest22/t2 (added), + pbx/ael/ael-test/ael-ntest22/t3 (added), + pbx/ael/ael-test/ael-ntest22/extensions.ael (added), + pbx/ael/ael-test/ael-ntest22 (added): This is a regression update + that matches what I did in 84134 for AEL regressions. + + * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes + 10786; All config files support #include with globbing (you know, + *,[chars],?,{list,list},etc), so I've updated the AEL system to + support this also. + +2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher + + * main/say.c: Correct pronunciations of numbers for .nl (Closes + issue #10837) + + * main/channel.c: Avoid a deadlock with ALL of the locks in the + masquerade function, not just the pairs of channels. (Closes + issue #10406) + +2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard + + * main/manager.c, channels/chan_agent.c, + include/asterisk/channel.h: if an Agent is redirected, the base + channel should actually be redirected. This was causing multiple + issues, especially issue 7706 and BE-160 + +2007-09-27 00:01 +0000 [r83976] Russell Bryant + + * pbx/pbx_dundi.c: remove a todo item that has been completed + +2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming + + * channels/chan_alsa.c: avoid the weird usage of assert() in the + ALSA header files that gcc 4.2 wants to complain about + +2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant + + * channels/chan_sip.c: I changed my mind ... I think this should be + a LOG_NOTICE. + + * channels/chan_sip.c: Add a log message that was requested by the + masses in the developer tutorial session at Astricon. chan_sip + did not output any message when a call was rejected because the + extension was not found. This adds a verbose message (at verbose + level 3) to note when this happens. + + * channels/chan_misdn.c: Fix building chan_misdn under dev-mode. + (please run the configure script with --enable-dev-mode so this + doesn't happen again ...) + +2007-09-26 18:35 +0000 [r83879] Tilghman Lesher + + * channels/chan_zap.c: Remove unused 4k of memory on the program + stack (closes issue #10827) + +2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher + + * main/app.c: jmls pointed out that unsetting the group and setting + the group to the blank string aren't quite the same. + + * build_tools/make_defaults_h: In the source, keys are relative to + the datadir, not varlib (which is the same in most cases, but + it's good to be accurate). Closes issue #10811 + + * doc/realtime.txt: Oops. Removed the unworkable workaround. This + note should never have been in the release. + + * main/app.c: Making change to group splitting, as discussed on the + -dev list. The main effect of this will be to permit + Set(GROUP([cat])=), i.e. unsetting a group. + +2007-09-24 07:54 +0000 [r83620] Christian Richter + + * channels/chan_misdn.c: fixed round_robin group dial method, this + never worked well on BRI Ports (2 channels) + +2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy + + * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes + are made here for the first arg in the for(arg1; arg2; arg3) {} + statement, as were done for the 3rd arg. It can now be an + assignment that will embedded in a Set() app, or a macro call, or + an app call. + + * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for + statement is now wrapped in Set() only if there's an '=' in that + string. Otherwise, if it begins with '&', then a Macro call is + generated; otherwise it is made into an app call. A bit more + accomodating, keeps the new guys happy, and the guys with ael-1 + code should be happy, too + +2007-09-21 14:37 +0000 [r83432] Russell Bryant + + * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c, + channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c, + main/file.c, include/asterisk/sched.h, channels/chan_h323.c, + pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl, + channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c, + main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y, + main/db1-ast/hash/hash.c, include/asterisk/channel.h, + channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing + with cosnt pointers. This set of changes gets all of Asterisk + (minus chan_alsa for now) to compile with gcc 4.2. (closes issue + #10774, patch from qwell) + +2007-09-21 13:34 +0000 [r83400] Joshua Colp + + * channels/chan_sip.c: Fix video under certain circumstances. It + would have been possible for the formats on the channel to not + contain the video format. (closes issue #10782) Reported by: + cwhuang + +2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant + + * main/asterisk.c: When daemonizing, don't change working directory + to "/". It makes it not be able to do a core dump when not + running as uid=root. (closes issue #10766, xrg) + + * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly + ask for /bin/bash, as it uses bashisms. (closes issue #10772, + reported by culrich) + +2007-09-20 17:09 +0000 [r83246] Jason Parker + + * apps/app_disa.c: If # is pressed after dialing an extension in + DISA, stop trying to collect more digits. (issue #10754) Reported + by: atis Patches: app_disa.c.branch.patch uploaded by atis + (license 242) app_disa.c.trunk.patch uploaded by atis (license + 242) + +2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp + + * channels/chan_sip.c: Make sure the minimum T1 timer value is + obeyed in all cases. (closes issue #10768) Reported by: flefoll + Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll + (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by + flefoll (license 244) + + * channels/chan_sip.c: Fix a minor spelling error. (closes issue + #10769) Reported by: flefoll Patches: + chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license + 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll + (license 244) + +2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant + + * apps/app_system.c: The System() and TrySystem() applications can + take a substantial amount of time to execute while not servicing + the channel. So, put the channel in autoservice while the command + is being executed. (closes issue #10726, reported by mnicholson) + + * funcs/func_curl.c: Using curl can take a substantial amount of + time, so the channel should be autoserviced while waiting for it + to complete. (closes issue #10725, reported by mnicholson) + + * channels/chan_iax2.c: When handling a reload of chan_iax2, don't + use an ao2_callback() to POKE all peers. Instead, use an + iterator. By using an iterator, the peers container is not locked + while the POKE is being done. It can cause a deadlock if the + peers container is locked because poking a peer will try to lock + pvt structs, while there is a lot of other code that will hold a + pvt lock when trying to go lock the peers container. (reported to + me directly by Loic Didelot. Thank you for the debug info!) + + * main/manager.c: Fix up another potential race condition. Do the + loop decrementing use count on events with the eventq protected + from being changed. (reported on IRC by Ivan) + +2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp + + * apps/app_queue.c: Protect the CDR record from modification by + pbx_exec so that the application data contains the Queue data. + (closes issue #10761) Reported by: snar Patches: + app-queue-mixmonitor.patch uploaded by snar (license 245) + + * channels/chan_sip.c: (closes issue #10760) Reported by: dimas + Patches: chan_sip.patch uploaded by dimas (license 88) Read in + subscribecontext option in general to be the default. + +2007-09-19 09:32 +0000 [r83023-83024] Christian Richter + + * channels/chan_misdn.c: removed comment which violates the coding + guidelines. + + * channels/misdn_config.c, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h: added 'astdtmf' option to + allow configuring the asterisk dtmf detector instead of the + mISDN_dsp ones. also added the patch from irroot #10190, so that + dtmf tones detected by the asterisk detector are passed outofband + to asterisk, to make any use of dtmf tones at all. + +2007-09-19 00:19 +0000 [r82992] Russell Bryant + + * apps/app_flash.c: Change the description of app_flash to note how + it can be a useful tool instead of just saying that it is + generally a worthless feature. (Thanks to Jim Van Meggelen for + pointing it out and providing the proposed text) + +2007-09-18 23:41 +0000 [r82961] Joshua Colp + + * apps/app_queue.c: Initialize a variable to NULL to make the world + happy. + +2007-09-18 22:42 +0000 [r82929] Russell Bryant + + * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle + interrupting the fgets() call when using FastAGI. This version of + the patch maintains the original behavior of the code when not + using FastAGI. (closes issue #10553) Reported by: juggie Patches: + res_agi_fgets-4.patch uploaded by juggie (license 24) + res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight + mods by me Tested by: juggie, festr + +2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey + + * main/manager.c: Corrected patch applied in revision r82887. + + * main/manager.c: Fixed a bug where http manager sessions prevented + the eventq from being cleaned out because http manager sessions + do not have a valid file descriptor. + +2007-09-18 20:56 +0000 [r82867] Russell Bryant + + * main/manager.c: Fix a memory leak that can occur on systems under + higher load. The issue is that when events are appended to the + master event queue, they use the number of active sessions as a + use count so it will know when all active sessions at the time + the event happened have consumed it. However, the handling of the + number of sessions was not properly synchronized, so the use + count was not always correct, causing an event to disappear + early, or get stuck in the event queue for forever. (closes issue + #9238, reported by bweschke, patch from Ivan, modified by me) + +2007-09-18 20:09 +0000 [r82865] Mark Michelson + + * apps/app_queue.c: Moving the logic for handling an empty + membername to the create_member function so that there is a + common place where this occurs instead of being spread out to + several different places. + +2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming + + * apps/app_queue.c: there is no need for conditional logic to + select ->interface or ->membername, snince ->membername will + always be populated + +2007-09-18 16:31 +0000 [r82802] Russell Bryant + + * pbx/pbx_dundi.c: When copying the contents from the wildcard + peer, do a deep copy instead of shallow copy so that it doesn't + crash when beging destroyed. (closes issue #10546, patch by me) + +2007-09-18 15:28 +0000 [r82751] Jason Parker + + * configs/sip.conf.sample: Correct the allowexternaldomains option + in SIP sample config. Issue 10753 + +2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant + + * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in + ast_localtime() instead of a couple places in app_voicemail to + prevent the problem everywhere instead of just a couple of + places. (related to issue #10746) + + * apps/app_voicemail.c: Initialize some memory to fix crashes when + leaving voicemail. This problem was fixed by running Asterisk + under valgrind. (closes issue #10746, reported by arcivanov, + patched by me) *** IMPORTANT NOTE: We need to check to see if + this same bug exists elsewhere. + + * res/res_features.c: Handle the case where there are multiple + dynamic features with the same digit mapping, but won't always + match the activated on/by access controls. In that case, the code + needs to keep trying features for a match. (reported by Atis on + the asterisk-dev list, patched by me) + +2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming + + * channels/chan_iax2.c: revert a change that wasn't supposed to be + committed... doh! + + * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places + where a logical member name (if specified) was not used, but + instead the direct interface was listed + +2007-09-17 02:00 +0000 [r82514] Joshua Colp + + * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead + of passing a NULL pointer into snprintf pass "". It makes Solaris + much happier. + +2007-09-14 21:19 +0000 [r82444] Steve Murphy + + * main/cdr.c: closes issue #10668; thanks to arkadia for his patch; + had to leave out the bit about ending the previous cdr in the + fork; it would destroy current implementations. + +2007-09-14 21:17 +0000 [r82435] Russell Bryant + + * configs/zapata.conf.sample: Add a note to help clarify the value + set with the echocancel option. (inspired by Malcolm's blog post + on blogs.digium.com about HPEC) + +2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson + + * apps/app_queue.c: Crap, I broke the build. Fixed. + + * apps/app_queue.c: Adding member name field to manager events + where they were missing before (closes issue #10721, reported by + snar) + +2007-09-14 17:48 +0000 [r82394] Jason Parker + + * channels/chan_zap.c: If a channel does not have an owner, do not + try to set a channel variable. This will end up making the + channel variable global, which is not right. Closes issue #10720, + patch by flefoll. + +2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + checking for libusb here, so nobody has to deal with conflicts in + the chan_usbradio-1.4 branch every time the configure script gets + changed + + * channels/chan_usbradio.c (removed), channels/xpmr (removed), + channels/Makefile: Remove chan_usbradio from the main 1.4 branch. + It can't live here because we have a strict policy to not include + new features in release branches. However, I'm going to merge it + into trunk, and I also have a special 1.4 based branch that + includes this module. svn co + http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4 + +2007-09-14 14:42 +0000 [r82376] Mark Michelson + + * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines + (closes issue #10717, reported and patched by leedm777) + +2007-09-14 01:24 +0000 [r82368] Jim Dixon + + * apps/app_rpt.c: Fixed problem with changes made to cdr + functionality + +2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming + + * channels/chan_usbradio.c: this new driver may not live in this + branch for long (since it is a new feature), but it definitely + should not be built by default + +2007-09-14 00:34 +0000 [r82366] Jim Dixon + + * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added), + channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added), + channels/xpmr (added), channels/xpmr/LICENSE (added), + channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample + (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added + channel driver for USB Radio device and support thereof. + +2007-09-13 23:11 +0000 [r82358] Jason Parker + + * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime + +2007-09-13 20:16 +0000 [r82346] Mark Michelson + + * apps/app_queue.c: Preemptively fixing a possible segfault. It is + possible that queuename is NULL (meaning pause ALL queues), so + use q->name instead. + +2007-09-13 20:11 +0000 [r82344] Jason Parker + + * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when + mutliple threads tried to close the same file. Do we actually + need the locking here? What happens if you open the same file + twice, and two threads try to write to it at the same time? Is + fputs() going to write out the entire line at once? I suspect + that it could be possible for the second fopen to run during the + first fputs, so the position could be in the middle of the + previously written line... Issue 10347, initial patch by + explidous (but I removed all of the paranoia stuff..) + +2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant + + * main/astobj2.c: resolve a warning when not building under dev + mode + + * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile + in tracking astobj2 statistics if dev-mode is enabled. Also, when + dev mode is enabled, register the CLI command that can be used to + run the astobj2 test and print out statistics. + +2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming + + * /, LICENSE: Merged revisions 82334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007) + | 2 lines clarify the OpenSSL and OpenH323 license exceptions + ........ + +2007-09-13 16:25 +0000 [r82326] Mark Michelson + + * apps/app_queue.c: Added logic to handle the unlikely case that + someone has two queues with the same name. Asterisk will log a + warning message letting the user know that one was already + defined with that name and is it skipping all further instances. + This also will work for realtime queues but in order for that to + happen, the user would have to trigger a perfectly timed reload + as a realtime queue is being looked up, which is highly unlikely + (but taken care of nonetheless). + +2007-09-13 11:47 +0000 [r82309] Philippe Sultan + + * channels/chan_gtalk.c: Closes issue #9401, reported and patched + by irrot, with slight modifications by me. Handle DTMF sent by + Asterisk properly. + +2007-09-12 21:56 +0000 [r82296] Russell Bryant + + * res/res_agi.c: Fix a check of the wrong pointer, as pointed out + by an XXX comment left in the code. The problem was harmless, + however. + +2007-09-12 21:28 +0000 [r82291] Tilghman Lesher + + * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone + files + +2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard + + * apps/app_meetme.c: remove a race condition for the creation of + recordthread's, and fix a small memory leak. This closes issue# + 10636 + +2007-09-12 20:12 +0000 [r82285] Tilghman Lesher + + * main/stdtime/private.h, main/stdtime/tzfile.h, + include/asterisk/localtime.h, main/stdtime/localtime.c: Working + on issue #10531 exposed a rather nasty 64-bit issue on + ast_mktime, so we updated the localtime.c file from source. Next + we'll have to write ast_strptime to match. + +2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant + + * main/asterisk.c: Clean up the output of "asterisk -h". This + tweaks the wording and wraps lines at 80 characters. (closes + issue #10699, seanbright) + + * res/res_agi.c: revert patch from issue #10553, as someone not + using fastagi reported that this broke their system. + +2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson + + * apps/app_voicemail.c: Accidentally committed changes to + app_voicemail which do NOT need to be in the 1.4 branch yet. + reverting... + + * apps/app_voicemail.c, apps/app_queue.c: We should only initialize + a realtime queue when it is allocated, not every time we access + it. This prevents the members ao2_container from being + reallocated every time the queue is accessed. I also removed a + debug message I had accidentally left in on a previous commit. + +2007-09-11 22:37 +0000 [r82267] Russell Bryant + + * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of + these calls was reading bogus memory ... + +2007-09-11 21:41 +0000 [r82265] Joshua Colp + + * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue + #10679) Reported by: andrew Build under dev mode when K6OPTS is + enabled. + +2007-09-11 20:49 +0000 [r82263] Russell Bryant + + * apps/app_queue.c: Fix another missing unref of member objects. + This one was pointed out by Marta. When building the outgoing + list in try_calling(), a member reference is stored in each + outgoing entry. However, when this list got destroyed, the + reference was not released. + +2007-09-11 20:36 +0000 [r82261] Steve Murphy + + * main/cdr.c: this change should fix issue # 10659 -- what I worry + about is how many other bug reports it may generate. Hopefully, + we can please the/a majority. Hopefully. We shall see. Calls not + marked ANSWERED and with only one channel name will not be + posted. This should eliminate the double CDR's. + +2007-09-11 16:05 +0000 [r82252] Mark Michelson + + * apps/app_queue.c: All instances of ao2_iterators which were just + named 'i' have been renamed to 'mem_iter' so that when refcounted + queues are merged into trunk, there will be little confusion + regarding iterator names, especially when a queue and member + iterator are used in the same function. + +2007-09-11 16:03 +0000 [r82250] Russell Bryant + + * pbx/pbx_dundi.c: The sample dundi.conf claims support for a + wildcard peer entry - [*], but the code did not support it. This + patch makes it work. (closes issue #10546, patch by dds, with + some changes by me) + +2007-09-11 16:01 +0000 [r82249] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a + hold/retrieve issue. + +2007-09-11 15:26 +0000 [r82245] Russell Bryant + + * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches: + res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: + juggie When using fastagi, fgets() can return before a full line + is read. Add explicit handling for the case where it gets + interrupted. + +2007-09-11 14:56 +0000 [r82243] Joshua Colp + + * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich + Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich + (license 176) Don't leak memory when unloading DUNDi. + +2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant + + * apps/app_queue.c: Add a couple more missing unrefs of queue + member objects + + * apps/app_queue.c: Add a missing unref of a queue member in an + error handling block + + * apps/app_queue.c: Document why membercount can not simply be + replaced by ao2_container_count() + + * main/astobj2.c: backport astobj2 race condition fix. This + function is the exact same as trunk so it applies here as well. + +2007-09-10 18:02 +0000 [r82155] Tilghman Lesher + + * apps/app_queue.c: Convert struct member to use refcounts (closes + issue #10199) + +2007-09-10 15:02 +0000 [r82091] Mark Michelson + + * configs/misdn.conf.sample: Removing non-existent options from + misdn configuration sample. (closes issue #10678, reported and + patched by IgorG) + +2007-09-09 02:35 +0000 [r82028] Tilghman Lesher + + * include/asterisk/lock.h: Fix inline compiles on really old + compilers (who uses gcc 2.7 anymore, really?) + +2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant + + * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit() + did not free the entry it removed. + + * .cleancount: (closes issue #10672) Bump the cleancount so that a + "make clean" will be forced. This is needed because my fix in + revision 81599 made a change to a data structure in file.h, and + since file dependency tracking is only on with dev-mode enabled, + file format modules that don't get rebuilt may crash, as is the + case with this issue. This makes me wonder - how much faster does + the code build without the file dependency tracking enabled? If + it doesn't make much of a difference, then it may be worth just + keeping it on all of the time, or perhaps just not in release + tarballs, so that this type of issue is avoided. + +2007-09-07 19:48 +0000 [r81923] Jason Parker + + * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used + as the mixmonitor filename. This moves the setting of the + MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch + by sim. + +2007-09-07 15:25 +0000 [r81886] Mark Michelson + + * configs/queues.conf.sample: Moving the explanation for joinempty + to a more appropriate place + +2007-09-06 22:28 +0000 [r81832] Russell Bryant + + * channels/chan_sip.c: (closes issue #9724, closes issue #10374) + Reported by: kenw Patches: 9724.txt uploaded by russell (license + 2) Tested by: kenw, russell Resolve a deadlock that occurs when + doing a SIP transfer to parking. I come across this type of + deadlock fairly often it seems. It is very important to mind the + boundary between the channel driver and the core in respect to + the channel lock and the channel-pvt lock. Channel drivers lock + to lock the pvt and then the channel once it calls into the core, + while the core will do it in the opposite order. The way this is + avoided is by having channel drivers either release their pvt + lock while calling into the core, or such as in this case, + unlocking the pvt just long enough to acquire the channel lock. + +2007-09-06 22:05 +0000 [r81778-81826] Jason Parker + + * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS + for ASTLDFLAGS. This adds LDOPTS + + * include/asterisk/astobj2.h: This should fix a build issue that + people building against uClibc were seeing with the addition of + astobj2 + +2007-09-06 19:40 +0000 [r81776] Joshua Colp + + * apps/app_meetme.c: (closes issue #10122) Reported by: + stevefeinstein Patches: meetme-unmute-manager.diff uploaded by + qwell (license 4) Tested by: stevefeinstein After looking over + the code I agree with Qwell. Setting the file descriptor to + conference each time just causes a fight back and forth. + +2007-09-06 16:56 +0000 [r81743] Philippe Sultan + + * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string + length fixes. Removed an unused variable in aji_client structure + (context) + +2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson + + * apps/app_queue.c: Fixes an issue where valid DTMF had to be + pressed twice to exit a queue if a member's phone was ringing. + (closes issue #10655, reported by strider2k, patched by me) + + * res/res_features.c: Fixes a memory leak (closes issue #10658, + reported and patched by Ivan) + +2007-09-06 14:20 +0000 [r81650] Philippe Sultan + + * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 - + and Google's XMPP server complaint, if set, the 'from' attribute + must be set to the user's full JID. + +2007-09-05 20:53 +0000 [r81599] Russell Bryant + + * include/asterisk/file.h, main/say.c, res/res_features.c, + main/file.c, include/asterisk/channel.h: Fix an issue that can + occur when you do an attended transfer to parking. If you + complete the transfer before the announcement of the parking spot + finishes, then the channel being parked will hear the remainder + of the announcement. These changes make it so that will not + happen anymore. Basically, res_features sets a flag on the + channel is playing the announcement to so that the file streaming + core knows that it needs to watch out for a channel masquerade, + and if it occurs, to abort the announcement. (closes BE-182) + +2007-09-05 17:18 +0000 [r81569] Tilghman Lesher + + * include/asterisk/lock.h: Solaris x86 compatibility fix + +2007-09-05 15:19 +0000 [r81525] Mark Michelson + + * apps/app_queue.c: Fixing the build... + +2007-09-05 15:14 +0000 [r81523] Jason Parker + + * channels/chan_phone.c: Do not try to unregister a NULL channel + tech. Also changed load_module function to use defines rather + than numbers for return values. Issue 10651, patch by + rbraun_proformatique, with additions by me. + +2007-09-05 15:03 +0000 [r81520] Mark Michelson + + * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to + only count members who are logged in and available. (related to + issue #10652, reported by wuwu) + +2007-09-05 13:11 +0000 [r81492] Joshua Colp + + * main/channel.c: (closes issue #10650) Reported by: tacvbo Only + print out that the spy was removed while holding the spy lock. + +2007-09-04 20:54 +0000 [r81453-81455] Jason Parker + + * apps/app_followme.c: Rather than attempt to play a file, we can + just check whether it exists. Issue 10634, patch by me, testing + by pabelanger, sanity checked by bweschke + + * configs/followme.conf.sample: Change default followme config file + to point to the correct files. Issue 10644, patch by pabelanger + +2007-09-04 18:37 +0000 [r81448] Russell Bryant + + * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: + Remove the typedefs on ao2_container and ao2_iterator. This is + simply because we don't typedef objects anywhere else in + Asterisk, so we might as well make this follow the same + convention. + +2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming + + * channels/chan_sip.c: there is no point in sending 401 + Unauthorized to a UAS that sent us a properly-formatted + Authentication header with the expected username and nonce but an + incorrect response (which indicates the shared secret does not + match)... instead, let's send 403 Forbidden so that the UAS + doesn't retry with the same authentication credentials repeatedly + +2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp + + * channels/chan_iax2.c: (closes issue #10632) Reported by: + jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded + by jamesgolovich (license 176) Fix memory leak when unloading + chan_iax2. The firmware files were not being freed. + + * main/channel.c: (closes issue #10476) Reported by: mdu113 Only + look for the end of a digit when waiting for a digit. This in + turn disables emulation in the core. + + * main/dns.c: (closes issue #10610) Reported by: john Patches: + dns.c.patch uploaded by john (license 218) Tested by: mvanbaak + Don't return a match if no SRV record actually exists. + +2007-09-03 18:57 +0000 [r81433] Russell Bryant + + * channels/chan_iax2.c: Remove a couple of calls to + ast_string_field_free_pools() on peers in error handling blocks + in the code for building peers. The peer object destructor does + this and doing it twice will cause a crash. (closes issue #10625, + reported by and patched by pnlarsson) + +2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson + + * apps/app_queue.c: Changed a comment to be more accurate. (really + this is just a test to make sure I can commit properly from home) + + * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr + into ao2_match_by_addr and making it available everywhere since + it could be a handy callback to have + +2007-08-31 21:27 +0000 [r81418] Russell Bryant + + * include/asterisk/astobj2.h: Remove references to a debugging + parameter that does not exist + +2007-08-31 19:48 +0000 [r81416] Mark Michelson + + * apps/app_queue.c: Fixed broken behavior of a reload on realtime + queues. Prior to this patch, if a reload was issued and a + realtime queue had callers waiting in it, then the queue would be + removed from the queue list, but it would not actually be freed + (in fact, a debug message warning about a memory leak would come + up). With this patch, reloads do not touch realtime queues at + all. + +2007-08-31 19:16 +0000 [r81415] Tilghman Lesher + + * funcs/func_logic.c: The IF() function was not allowing true + values that had embedded colons (closes issue #10613) + +2007-08-31 18:44 +0000 [r81412] Jason Parker + + * apps/app_dial.c: Re-order dial options to be in line with the + existing alpha order. Issue 10621, initial patch by junky + +2007-08-31 17:38 +0000 [r81410] Philippe Sultan + + * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command + available. Closes issue 10548, reported by keepitcool. + +2007-08-31 15:53 +0000 [r81406] Joshua Colp + + * res/res_speech.c: Make it the engine's responsible to check for + the presence of results. + +2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming + + * codecs/codec_zap.c: add missing "transcoder show" (and deprecated + "show transcoder") CLI commands that were in 1.2 but never added + to 1.4 + +2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp + + * res/res_features.c: (closes issue #10618) Reported by: dimas + Don't pass through the stopped sounds frame.... just drop it. + + * res/res_features.c: (closes issue #10009) Reported by: dimas + Don't output a bridge failed warning message if it failed because + one of the channels was part of the masquerade process. That is + perfectly normal. + +2007-08-30 22:05 +0000 [r81397] Mark Michelson + + * apps/app_queue.c: Removing an extraneous (and possibly + misleading) log message. Firstly, if the announce file isn't + found, the streaming functions will report it. Secondly, not all + non-zero returns from play_file mean that the announce file + wasn't found. Positive return values simply mean that a digit was + pressed (most likely to skip through the announcement). (closes + issue #10612, reported and patched by dimas) + +2007-08-30 21:23 +0000 [r81395] Joshua Colp + + * channels/chan_sip.c: (closes issue #10514) Reported by: casper + Patches: chan_sip.c.80129.diff uploaded by casper (license 55) + Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible + for it to ever be that value. + +2007-08-30 21:11 +0000 [r81392] Steve Murphy + + * main/cdr.c: via issue 10599, where 'CDR already initialized' + messages are being generated. Since all channels will have an + init'd CDR attached at creation time, this message is now + particularly useless. Removed. + +2007-08-30 15:38 +0000 [r81383] Russell Bryant + + * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING + define. (closes issue #10559, paravoid) + +2007-08-30 15:35 +0000 [r81381] Mark Michelson + + * apps/app_queue.c: Changed some manager event messages to reflect + whether a queue member is a realtime member or not + +2007-08-30 15:33 +0000 [r81379] Russell Bryant + + * configs/modem.conf.sample (removed), configs/enum.conf.sample, + configs/extensions.ael.sample: Fix a typo, update a reload + command, and remove an unused configuration file. (closes issue + #10606, casper) + +2007-08-30 14:53 +0000 [r81375] Joshua Colp + + * main/pbx.c: (closes issue #10603) Reported by: jmls Patches: + pbx.diff uploaded by jmls (license 141) Backport changes from + 81372. Add REASON dialplan variable for when an originated call + fails and the failed extension is executed. + +2007-08-30 14:43 +0000 [r81373] Christian Richter + + * channels/chan_misdn.c: Fixed some warnings. + +2007-08-30 14:23 +0000 [r81369] Joshua Colp + + * res/res_features.c: (issue #10599) Reported by: dimas Handle the + -1 control subclass during feature dialing (it indicates to stop + sounds). + +2007-08-30 08:31 +0000 [r81367] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe + issue where a misdn_read would lock the channel, but read would + not return because it blocks. later chan_misdn would try to queue + a frame like a AST_CONTROL_ANSWER which could result in a + deadlock situation. misdn_read will now not block forever + anymore, and we don't queue the ANSWER frame at all when we + already was called with misdn_answer -> answer would be called + twice. Also we don't explicitly send a RELEASE_COMPLETE on + receiption of a RELEASE anymore, because mISDN does that for us, + this resulted in a problem on some switches, which would block + our port after some calls for a short while. + +2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson + + * apps/app_queue.c: This patch, in essence, will correctly pause a + realtime queue member and reflect those changes in the realtime + engine. (issue #10424, reported by irroot, patch by me) This + patch creates a new function called update_realtime_member_field, + which is a generic function which will allow any one field of a + realtime queue member to be updated. This patch only uses this + function to update the paused status of a queue member, but it + lays the foundation for persisting the state of a realtime member + the same way that static members' state is maintained when using + the persistentmembers setting + + * apps/app_queue.c: Changed some tabs to spaces + +2007-08-29 15:57 +0000 [r81342] Russell Bryant + + * main/Makefile: If chan_h323 is not being built, don't use g++ to + do the final link of Asterisk. (in response to a question on the + asterisk-dev list) + +2007-08-29 15:52 +0000 [r81340] Mark Michelson + + * apps/app_queue.c: This fix creates a more accurate way of + detecting whether realtime members were deleted. (closes issue + 10541, reported by Alric, patched by me) The REALLY nice things + about this patch is that queue members now have a "realtime" + field which will be true if the member is a realtime member. This + means we can check this value prior to certain processing if it + should ONLY be done for realtime members. + +2007-08-29 14:13 +0000 [r81331] Joshua Colp + + * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make + rtp timeouts work even if two RTP streams are directly bridged in + the RTP stack. + +2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant + + * channels/chan_iax2.c: Change the message about receiving a + mini-frame before the first full voice frame to a DEBUG message. + + * pbx/pbx_dundi.c: revert unintentional changes in rev 81226 + + * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian + tones. (closes issue #7953, hanabana) + +2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson + + * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using + res_config_mysql (closes issue #10573, reported by chrisvaughan, + patch suggested by chrisvaughan as well) + + * apps/app_queue.c: Resolve a potential deadlock. In this case, a + single queue is locked, then the queue list. In changethread(), + the queue list is locked, and then each individual queue is + locked. Under the right circumstances, this could deadlock. As + such, I have unlocked the individual queue before locking the + queue list, and then locked the queue back after the queue list + is unlocked. + + * channels/chan_agent.c: DTMF begin frames should be ignored so + that when an agent acks a call with the '#' key, he doesn't cause + a queue's announce file to be interrupted. Also went ahead and + did the same for the '*' key and for ending a call. (closes issue + #10528, reported by deskhack, patched by me) + +2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant + + * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most + of the memory it allocates on unload and should be fixed (when + I'm not in the middle of something else ...). + + * pbx/pbx_dundi.c: explicity define a variable as a boolean + + * res/res_musiconhold.c: (closes issue #10419) Reported by: + mustardman Patches: asterisk-mohposition.diff.txt uploaded by + jamesgolovich (license 176) This patch fixes a few problems with + music on hold. * Fix issues with starting at the beginning of a + file when it shouldn't. * Fix the inuse counter to be decremented + even if the class had not been set to be deleted when not in use + anymore * Don't arbitrarily limit the number of MOH files to 255 + +2007-08-27 15:01 +0000 [r81012] Joshua Colp + + * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang + Patches: chan_sip-ChannelReload-20080825.patch uploaded by + jesselang (license 202) Remove an extra \r\n to make the + ChannelReload event conform with every other event. + +2007-08-27 14:55 +0000 [r81010] Mark Michelson + + * apps/app_queue.c: Found a case where the queue's membercount is + off. It does not take into account dynamic members on a reload. + +2007-08-27 13:20 +0000 [r80974] Joshua Colp + + * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct + jitter value output in the CLI to be as expected. + +2007-08-26 18:11 +0000 [r80932] Russell Bryant + + * channels/chan_iax2.c: Remove an extra signal_condition() for the + scheduler thread. (closes issue #10564, patch from casper) + +2007-08-25 17:37 +0000 [r80895] Russell Bryant + + * channels/chan_iax2.c: Fix some issues with the handling of the + scheduler in chan_iax2. Most of the places that scheduled items + to be executed by the scheduler thread did not signal the + scheduler thread to wake up so that it could recalculate the time + until the next action. These changes will make the scheduler + thread more responsive and ensure that actions get executed as + close to when intended as possible instead of it being possible + for very long delays. + +2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard + + * apps/app_zapateller.c: An empty string is an empty callerid ... + so zap it. This closes issue #10502, which was pointed out by + dswartz. Thank you, and may the swartz be with you + +2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant + + * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are + available when Asterisk first starts, then don't give up on + poking peers. Allow the poke to get rescheduled so that it will + work once the dnsmgr is able to resolve the host. (closes issue + #10521, patch by jamesgolovich) + + * main/dsp.c: Improve the debouncing logic in the DTMF detector to + fix some reliability issues. Previously, this code used a shift + register of hits and non-hits. However, if the start of the digit + isn't clean, it is possible for the leading edge detector to miss + the digit. These changes replace the flawed shift register logic + and also does the debouncing on the trailing edge as well. + (closes issue #10535, many thanks to softins for the patch) + +2007-08-24 19:52 +0000 [r80818] BJ Weschke + + * apps/app_queue.c: A minor correction to the available logic of + autofill. If a queue member is paused, they're not really + "available" so don't count them as such. Somewhat related to + issue #10155 + +2007-08-24 18:52 +0000 [r80789] Steve Murphy + + * main/cdr.c: From a complaint by jmls, I realize that the message + in cdr_disposition is unnecessary. To get failure disposition, + just return -1; no use having more than one case do that. + +2007-08-24 15:51 +0000 [r80750] Mark Michelson + + * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail. + +2007-08-24 15:41 +0000 [r80747] Tilghman Lesher + + * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with + the code, instead of only in documentation (closes issue #10549) + +2007-08-24 15:28 +0000 [r80722] Russell Bryant + + * utils/ael_main.c: Tweak the formatting of this MODULEINFO block. + I think this would have caused a "*" to get in the + menuselect-tree file. + +2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy + + * utils/ael_main.c: This change addresses JerJer's complaint that + aelparse builds and installs even if pbx_ael is unchecked in the + menuselect stuff. + + * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6: + backport of 80649, a fix to an unreported problem in the ael + parser, that results in a crash on a 64bit machine + +2007-08-24 11:42 +0000 [r80661] Philippe Sultan + + * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are + answered too early, which results in CDRs wrongly stating that a + call was ANSWERED when the calling party cancelled a call before + before being established. We must not answer the call upon + reception of a 'transport-accept' iq packet, but this packet + still needs to be acknowledged, otherwise the remote peer would + close the call (like in #8970). + +2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard + + * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without + warnings + + * channels/chan_misdn.c: make chan_misdn compile without warnings + +2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant + + * include/asterisk/features.h, res/res_features.c: When executing a + dynamic feature, don't look it up a second time by digit pattern + after we already looked it up by name. This causes broken + behavior if there is more than one feature defined with the same + digit pattern. (closes issue #10539, reported by bungalow, patch + by me) + + * funcs/func_timeout.c: Revert very broken fix for issue #10540 ... + none of these values take ms so I don't know what I was thinking + + * funcs/func_timeout.c: Fix func_timeout to take values in floating + point so 1.5 actually means 1.5 seconds instead of being rounded. + (closes issue #10540, reported by spendergrass, patch by me) + +2007-08-23 17:14 +0000 [r80505-80507] Jason Parker + + * /: *sigh* + + * /: use autotagged externals + +2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming + + * channels/chan_zap.c: report the actual channel number that was + unregistered, instead of assuming that the interface list + consists of channels 1 through with no gaps in the sequence + +2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant + + * channels/chan_iax2.c: Fix some code where it was possible for a + reference to a peer to not get released when it should. Thank you + to Marta Carbone for pointing this out! + + * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c: + This is a hack to maintain old behavior of chan_iax2. This + ensures that if the peers and users are being stored in a linked + list, that they go in the list in the same order that the older + code used. This is necessary to maintain the behavior of which + peers and users get matched when traversing the container. + + * res/res_agi.c: Revert res_agi fix that didn't quite work until we + get it right ... + + * include/asterisk/astobj2.h: Add some more documentation on + iterating ao2 containers. The documentation implies that is + possible to miss an object or see an object twice while + iterating. After looking through the code and talking with + mmichelson, I have documented the exact conditions under which + this can happen (which are rare and harmless in most cases). + + * main/astobj2.c: When converting this code to use the list macros, + I changed it so objects are added to the head of a bucket instead + of the tail. However, while looking over code with mmichelson, we + noticed that the algorithm used in ao2_iterator_next requires + that items are added to the tail. This wouldn't have caused any + huge problem, but it wasn't correct. It meant that if an object + was added to a container while you were iterating it, and it was + added to the same bucket that the current element is in, then the + new object would be returned by ao2_iterator_next, and any other + objects in the bucket would be bypassed in the traversal. + + * channels/chan_sip.c: Don't crash when using realtime in chan_sip + without an insecure setting in the database. (closes issue + #10348, reported by link55, fixed by me) + + * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h + (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h, + channels/chan_iax2.c: Merge changes from + team/russell/iax_refcount. This set of changes fixes problems + with the handling of iax2_user and iax2_peer objects. It was very + possible for a thread to still hold a reference to one of these + objects while a reload operation tries to delete them. The fix + here is to ensure that all references to these objects are + tracked so that they can't go away while still in use. To + accomplish this, I used the astobj2 reference counted object + model. This code has been in one of Luigi Rizzo's branches for a + long time and was primarily developed by one of his students, + Marta Carbone. I wanted to go ahead and bring this in to 1.4 + because there are other problems similar to the ones fixed by + these changes, so we might as well go ahead and use the new + astobj if we're going to go through all of the work necessary to + fix the problems. As a nice side benefit of these changes, peer + and user handling got more efficient. Using astobj2 lets us not + hold the container lock for peers or users nearly as long while + iterating. Also, by changing a define at the top of chan_iax2.c, + the objects will be distributed in a hash table, drastically + increasing lookup speed in these containers, which will have a + very big impact on systems that have a large number of users or + peers. The use of the hash table will be made the default in + trunk. It is not the default in 1.4 because it changes the + behavior slightly. Previously, since peers and users were stored + in memory in the same order they were specified in the + configuration file, you could influence peer and user matching + order based on the order they are specified in the configuration. + The hash table does not guarantee any order in the container, so + this behavior will be going away. It just means that you have to + be a little more careful ensuring that peers and users are + matched explicitly and not forcing chan_iax2 to have to guess + which user is the right one based on secret, host, and access + list settings, instead of simply using the username. If you have + any questions, feel free to ask on the asterisk-dev list. + + * res/res_agi.c: Juggie in #asterisk-dev was reporting problems + where fgets would return without reading the whole line when + using fastagi. When this happens, errno was set to EINTR or + EAGAIN. This patch accounts for the possibility and lets fgets + continue in that case. + +2007-08-22 18:53 +0000 [r80302-80330] Jason Parker + + * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h, + build_tools/strip_nonapi, build_tools/prep_moduledeps, + build_tools/make_buildopts_h: Fix a few build issues in Solaris + (and likely others). Use GREP and ID variables from autoconf. + Reported to me in #asterisk-dev I forgot who reported this - + sorry. :( + + * Makefile: Change a syntax that the GNU make in Solaris dislikes. + + * build_tools/make_version: Fix a bashism (we explicitly request + /bin/sh). Remove some oddly placed quotes I found in passing. + +2007-08-22 16:21 +0000 [r80257] Russell Bryant + + * Makefile: Honor the contents of the COPTS variable as custom + target CFLAGS. Apparently this is what openwrt does. (reported by + Brian Capouch on the asterisk-dev list, patch by me) + +2007-08-22 16:14 +0000 [r80255] Joshua Colp + + * main/rtp.c: (closes issue #10526) Reported by: sinistermidget + Revert commit from issue #10355 and return timestamp skew to 640. + +2007-08-21 Russell Bryant + + * Asterisk 1.4.11 released. + +2007-08-21 18:42 +0000 [r80183] Russell Bryant + + * channels/chan_sip.c: Don't record SIP dialog history if it's not + turned on. Also, put an upper limit on how many history entires + will be stored for each SIP dialog. It is currently set to 50, + but can be increased if deemed necessary. (closes issue #10421, + closes issue #10418, patches suggested by jmoldenhauer, patches + updated by me) (Security implications documented in AST-2007-020) + +2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy + + * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing + the diffs from ulaw.h and alaw.h for now; accidentally added them + in 80166 + + * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h: + This patch solves problem 1 in 8126; it should not slow down the + alaw codec, but should prevent signal degradation via multiple + trips thru the codec. Fossil estimates the twice thru this codec + will prevent fax from working. 4-6 times thru would result + hearable, noticeable, voice degradation. + +2007-08-21 15:22 +0000 [r80132] Russell Bryant + + * channels/chan_mgcp.c: Don't try to dereference the owner channel + when it may not exist (issue #10507, maxper) + +2007-08-21 15:03 +0000 [r80130] Jason Parker + + * configs/cdr.conf.sample: (issue #10510) Reported by: casper + Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few + errors in sample cdr config file. + +2007-08-20 21:57 +0000 [r80088] Russell Bryant + + * apps/app_queue.c: Fix the build of app_queue + +2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson + + * apps/app_queue.c: After a discussion on #asterisk-dev, it was + decided that this should be in 1.4 as well. (issue #10424, + reported and patched by irroot) + + * apps/app_queue.c: Found a pointless ternary if. member->dynamic + was set to 1 and has no opportunity to change between then and + this line, so "dynamic" will ALWAYS be output. + +2007-08-20 16:08 +0000 [r80047] Jason Parker + + * configs/extensions.conf.sample: (issue #10499) Reported by: + casper Patches: extensions.conf.sample.diff uploaded by casper + (license 55) Update CLI examples in extensions.conf.sample to + reflect command changes. + +2007-08-20 15:34 +0000 [r80044] Mark Michelson + + * apps/app_voicemail.c: Ukrainian language voicemail support. + (closes issue #10458, reported and patched by Oleh) + +2007-08-20 02:42 +0000 [r79998] Tilghman Lesher + + * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by + snuffy via IRC) + +2007-08-18 14:30 +0000 [r79947] Tilghman Lesher + + * apps/app_voicemail.c: Don't allocate vmu for messagecount when we + could just use the stack instead (closes issue #10490) Also, + remove a useless (and leaky) SQLAllocHandle (closes issue #10480) + +2007-08-17 21:01 +0000 [r79912] Russell Bryant + + * channels/chan_zap.c: Avoid a crash in the handling of DTMF based + Caller ID. It is valid for ast_read to return NULL in the case + that the channel has been hung up. (crash reported by + anonymouz666 on IRC in #asterisk-dev) + +2007-08-17 19:14 +0000 [r79906] Mark Michelson + + * apps/app_voicemail.c: Patch allows for more seamless transition + from file storage voicemail to ODBC storage voicemail. If a + retrieval of a greeting from the database fails, but the file is + found on the file system, then we go ahead an insert the greeting + into the database. The result of this is that people who switch + from file storage to ODBC storage do not need to rerecord their + voicemail greetings. + +2007-08-17 19:12 +0000 [r79902-79904] Jason Parker + + * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: + Don't send a semicolon over the wire in sip notify messages. + Caused by fix for issue 9938. I basically took the code that + existed before 9938 was fixed, and copied it into a new function + - ast_unescape_semicolon There should be very few places this + will be needed (pbx_config does NOT need this (see issue 9938 for + details)) Issue 10430, patch by me, with help/ideas from murf + (thanks murf). + + * channels/chan_local.c: Re-add the setting of callerid name and + number. Issue 10485, reported by and fix explained by paradise. + +2007-08-17 13:37 +0000 [r79857] Russell Bryant + + * channels/chan_sip.c: Fix some crashes in chan_sip. This patch + changes various places that add items to the scheduler to ensure + that they don't overwrite the ID of a previously scheduled item. + If there is one, it should be removed. (closes issue #10391, + closes issue #10256, probably others, patch by me) + +2007-08-17 08:22 +0000 [r79833] Christian Richter + + * channels/chan_misdn.c: sometimes we don't need to signal dtmf + tones to asterisk, we just want them to go through as inband. + Otherwise they might be generated by the other channel partner + and then there is a double tone. + +2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant + + * res/res_musiconhold.c: Fix a little race condition that could + cause a crash if two channels had MOH stopped at the same time + that were using a class that had been marked for deletion when + its use count hits zero. + + * res/res_musiconhold.c: This patch fixes a bug where reloading the + module with "module reload" did not delete classes from memory + that were no longer in the config. This patch fixes that problem + as well as another one. Previously, if you reloaded MOH using the + "moh reload" CLI command, which behaved differently than "module + reload ...", MOH had to be stopped on every channel and started + again immediately. However, there was no way to tell what class + was being used, so they would all fall back to the default class. + (closes issue #10139) Reported by: blitzrage Patches: + asterisk-10139-advanced.diff.txt uploaded by jamesgolovich + (license 176) Tested by: jamesgolovich + + * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were + introduced by making frame handling and scheduling + multi-threaded. Unfortunately, we have to do some expensive + deadlock avoidance when queueing frames on to the ast_channel + owner of the IAX2 pvt struct. This was already handled for + regular frames, but ast_queue_hangup and ast_queue_control were + still used directly. Making these changes introduced even more + places where the IAX2 pvt struct can disappear in the context of + a function holding its lock due to calling a function that has to + unlock/lock it to avoid deadlocks. I went through and fixed all + of these places to account for this possibility. (issue #10362, + patch by me) + +2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson + + * channels/chan_agent.c: Fixes a problem where agents would get + stuck busy due to their wrapuptime being longer than the queue's + wrapuptime and ringinuse=no for the queue. (closes issue #10215, + reported by Doug, repaired by me) Special thanks to fkasumovic + for pointing out the source of the problem and to bweschke for + helping to come up with a solution! + + * apps/app_voicemail.c: base_encode is not trying to open a log + file, so we should not call it a log file in the warning. + (related to issue #10452, reported by bcnit) + +2007-08-16 09:37 +0000 [r79665] Philippe Sultan + + * res/res_jabber.c: A fix for two critical problems detected while + working with Daniel McKeehan in issue #10184. Upon priority + change, the resource list is not NULL terminated when moving an + item to the end of the list. This makes Asterisk endlessy loop + whenever it needs to read the list. Jids with different resource + and priority values, like in Gmail's and GoogleTalk's jabber + clients put that problem in evidence. Upon reception of a 'from' + attribute with an empty resource string, Asterisk crashes when + trying to access the found->cap pointer if the resource list for + the given buddy is not empty. This situation is perfectly valid + and must be handled. The Gizmoproject's jabber client put that + problem in evidence. Also added a few comments in the code as + well as a handle for the capabilities from Gmail's jabber client, + which are stored in a caps:c tag rather than the usual c tag. + Closes issue #10184. + +2007-08-16 08:21 +0000 [r79642] Christian Richter + + * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when + we want to send IA5 Chars. + +2007-08-15 14:40 +0000 [r79553] Joshua Colp + + * main/rtp.c: (closes issue #10440) Reported by: irroot (closes + issue #10454) Reported by: flo_turc Increase maximum timestamp + skew to 120. 20 was apparently far too low. + +2007-08-15 14:26 +0000 [r79527] Mark Michelson + + * apps/app_voicemail.c: Fixed an error in the Russian language + voicemail intro. (issue #10458, reported and patched by Oleh) + +2007-08-15 14:18 +0000 [r79523] Joshua Colp + + * channels/chan_sip.c: (closes issue #10456) Reported by: irroot + Patches: sip_timeout.patch uploaded by irroot (license 52) Change + hardcoded timer value to defined value. I'm doing this in 1.4 as + well so if it needs to be changed in the future this place would + not have been forgotten. + +2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant + + * channels/chan_iax2.c: Fix another spot where an iax2_peer would + be leaked if realtime was in use. + + * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2 + related to the use of realtime. I found these while working on + iax2_peer object reference tracking. + +2007-08-14 15:27 +0000 [r79397] Joshua Colp + + * res/res_features.c: (closes issue #10415) Reported by: atis + Revert fix for #10327 as it causes more issues then it solves. + +2007-08-13 22:40 +0000 [r79363] Steve Murphy + + * pbx/pbx_ael.c: memset really, really needs to be used here. + +2007-08-13 21:57 +0000 [r79334] Joshua Colp + + * res/res_speech.c, apps/app_speech_utils.c, + include/asterisk/speech.h: Instead of accepting a single DTMF + character accept a full string. + +2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant + + * channels/chan_iax2.c: Don't call find_peer in + registry_authrequest with the pvt lock held to avoid a deadlock. + + * channels/chan_iax2.c: Release the pvt lock before calling + find_peer in register_verify to avoid a deadlock. Also, remove + some unnecessary locking in auth_fail that was only done + recursively. + + * channels/chan_iax2.c: Don't call find_peer within update_registry + with a pvt lock held. This can cause a deadlock as the code will + eventually call find_callno. + + * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I + have tracked them down to a single core issue. You can not call + find_callno() while holding a pvt lock as this function has to + lock another (every) other pvt lock. Doing so can lead to a + classic deadlock. So, I am tracking down all of the code paths + where this can happen and fixing them. The fix I committed + earlier today was along the same theme. This patch fixes some + code down the path of authenticate_reply. + +2007-08-13 17:49 +0000 [r79255] Steve Murphy + + * pbx/ael/ael-test/ref.ael-vtest21 (added), + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ael-vtest21/extensions.ael (added), + pbx/ael/ael-test/ael-vtest21 (added), + pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c, + pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch + fixes bug 10411. I added a new regression test, some regression + test cleanups + +2007-08-13 15:28 +0000 [r79214] Russell Bryant + + * channels/chan_iax2.c: Fix a potential deadlock in socket_process. + check_provisioning can eventually call find_callno. You can't + hold a pvt lock while calling find_callno because it goes through + and locks every single one looking for a match. + +2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp + + * res/res_speech.c, apps/app_speech_utils.c, + include/asterisk/speech.h: Add an API call to allow the engine to + know that DTMF was received. + + * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c, + channels/chan_local.c, channels/chan_misdn.c, + channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_h323.c, channels/chan_gtalk.c, + channels/chan_iax2.c: (closes issue #10437) Reported by: haklin + Don't set the callerid name and number a second time on a newly + created channel. ast_channel_alloc itself already sets it and + setting it twice would cause a memory leak. + +2007-08-11 05:23 +0000 [r79142] Tilghman Lesher + + * res/res_odbc.c: Ensure the connection gets marked as used at + allocation time (closes issue #10429, report and fix by + mnicholson) + +2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy + + * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From + a user complaint on #asterisk, I have forced pbx_spool to explain + what reason codes mean, when they are logged + + * main/cdr.c: Re bug behavior mentioned in #asterisk, made this + tweak to code, to prevent hundreds of log messages from being + generated + + * main/cdr.c: This will help debug; from a question asked on + #asterisk + +2007-08-10 Russell Bryant + + * Asterisk 1.4.10.1 released. + +2007-08-10 15:20 +0000 [r78995] Russell Bryant + + * include/asterisk/lock.h: The last set of changes that I made to + "core show locks" made it not able to track mutexes unless they + were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized + with ast_mutex_init() were not tracked. It should work now. + +2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp + + * main/file.c: Don't bother having the core pass through or emulate + begin DTMF frames when in an ast_waitstream. It only cares about + the end of DTMF. + + * configs/queues.conf.sample: (closes issue #10422) Reported by: + bhowell Add note to sample configuration about module load order + and how it can cause perfectly good queue members to be marked as + invalid. + +2007-08-10 13:24 +0000 [r78936] Christian Richter + + * channels/chan_misdn.c, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser + information element. We send them now also in the disconnect + message. + +2007-08-09 23:47 +0000 [r78907] Mark Michelson + + * apps/app_voicemail.c: Improved a bit of logic regarding + comma-separated mailboxes in has_voicemail. Also added some + braces to some compound if statements since unbraced if + statements scare me in general. + +2007-08-09 23:10 +0000 [r78891] Steve Murphy + + * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty + output + +2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson + + * apps/app_voicemail.c: Removing some extra debug code I left in my + last commit + + * apps/app_voicemail.c: Quite a few changes regarding IMAP storage. + 1. instead of using inboxcount as the core message counting + function, we use messagecount instead. This makes it possible to + count messages in folders besides just INBOX and Old. 2. + inboxcount and hasvoicemail now use messagecount as their means + of determining return values. 3. Added a copy_message function + for IMAP storage. Unfortunately I don't have the means to test + it, but it seems like a pretty straightforward function. 4. + Removed a #ifndef IMAP_STORAGE and matching #endif from + leave_voicemail for a couple of reasons. One, we want to support + copying mail to multiple IMAP boxes, and two, IMAP was broken + because a STORE macro had been moved into this section of code. + + * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I + put some new code in the wrong place and so I've reverted the + canreinvite section to how it was and put my new code where it + should be. + +2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant + + * apps/app_voicemail.c: add a comment to indicate that inboxcount + for ODBC_STORAGE needs to be fixed to support multiple mailboxes + + * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for + ODBC_STORAGE. Also, leave a comment for this to be fixed for + IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was + working on this code right now for another reason. This is broken + even worse in trunk, but for a different reason. The fact that + the mailbox option supported multiple mailboxes is completely not + obvious from the code in the channel drivers. Anyway, I will fix + that in another commit ... + + * apps/app_meetme.c: Fix a problem with the combination of the 'F' + option to pass DTMF through a conference and options that use + DTMF to activate various features. The problem was that the BEGIN + frame would be passed through, but the END frame would get + intercepted to activate a feature. Then, the other conference + members would hear DTMF for forever, which they didn't seem to + like very much. (closes issue #10400, reported by stevefeinstein, + fixed by me) + +2007-08-08 19:29 +0000 [r78646] Jason Parker + + * doc/jabber.txt: Fix mogs email address. + +2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson + + * apps/app_voicemail.c: Fixed some compiler warnings so that + compiling with dev-mode and IMAP storage would not have any + errors. This section of code may get changed again shortly since + my change uncovers a rather silly bit of logic. + + * apps/app_queue.c: Changing a bit of logic so that someone will + NEVER exit the queue on timeout unless they have enabled the 'n' + option. This commit relates to issue #10320. Thanks to + jfitzgibbon for detailing the idea behind this code change. + +2007-08-08 13:51 +0000 [r78569] Joshua Colp + + * configs/sip.conf.sample: (closes issue #10335) Reported by: + adamgundy Update sip.conf to include another scenario where + directrtpsetup will fail. + +2007-08-07 Russell Bryant + + * Asterisk 1.4.10 released. + +2007-08-07 20:57 +0000 [r78488] Russell Bryant + + * res/res_config_odbc.c: Fix the build of this module on 64-bit + platforms + +2007-08-07 19:43 +0000 [r78450] Mark Michelson + + * apps/app_voicemail.c: The logic behind inboxcount's return value + was reversed in has_voicemail and message_count. (closes issue + #10401, reported by st1710, patched by me) + +2007-08-07 19:34 +0000 [r78437] Tilghman Lesher + + * res/res_odbc.c: Don't free the environment handle when the + connection fails, because other connections might be depending + upon it + +2007-08-07 19:11 +0000 [r78416] Jason Parker + + * channels/chan_sip.c: Allow chan_sip to build in devmode + +2007-08-07 19:09 +0000 [r78415] Tilghman Lesher + + * apps/app_voicemail.c, res/res_config_odbc.c, + apps/app_directory.c: Reconnection doesn't happen automatically + when a DB goes down (fixes issue #9389) + +2007-08-07 18:25 +0000 [r78375] Jason Parker + + * channels/chan_skinny.c: Properly check the capabilities count to + avoid a segfault. (ASA-2007-019) + +2007-08-07 17:45 +0000 [r78371] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | + 4 lines Revert patch committed for issue #9660. It broke E&M + trunks. (closes issue #10360) (closes issue #10364) ........ + +2007-08-06 21:41 +0000 [r78275] Joshua Colp + + * main/channel.c: Add additional DTMF log messages to help when + debugging issues. + +2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant + + * channels/chan_iax2.c: Fix an issue where dynamic threads can get + free'd, but still exist in the dynamic thread list. (closes issue + #10392, patch from Mihai, with credit to his colleague, Pete) + + * include/asterisk/linkedlists.h: Fix the return value of + AST_LIST_REMOVE(). This shouldn't be causing any problems, + though, because the only code that uses the return value only + checks to see if it is NULL. (closes issue #10390, pointed out by + mihai) + +2007-08-06 16:32 +0000 [r78182] Joshua Colp + + * channels/chan_sip.c: It is possible for a transfer to occur + before the remote device has our tag in which case they send none + in the transfer. In this case we need to not fail the transfer + dialog lookup. + +2007-08-06 16:30 +0000 [r78180] Jason Parker + + * main/config.c: Fix an issue with using UpdateConfig (manager + action) where escaped semicolons in a config would be converted + to just semicolons (\; to ;) Issue 9938 + +2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp + + * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that + we pass through RTP timestamp information we need to make the + allowed timestamp skew considerably less. There are situations + where a source may change and due to the timestamp difference the + receiver will experience an audio gap since we did not indicate + by setting the marker bit that the source changed. + + * configure, configure.ac: (closes issue #10383) Reported by: rizzo + Include stdlib.h so NULL gets defined for gethostbyname_r checks. + +2007-08-06 13:33 +0000 [r78164] Mark Michelson + + * channels/chan_sip.c: Fixed a mistake I made in realtime_peer + which caused it to return NULL every time. Thanks to Jon Fealy + for emailing me the correction. + +2007-08-05 14:18 +0000 [r78146] Tilghman Lesher + + * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes + bug #10382) + +2007-08-05 04:15 +0000 [r78143] Russell Bryant + + * include/asterisk/lock.h: Fix compilation failure when + MALLOC_DEBUG is enabled, but DEBUG_THREADS is not + +2007-08-05 03:29 +0000 [r78139] Tilghman Lesher + + * channels/chan_sip.c: If peer is not found, the error message is + misleading (should be peer not found, not ACL failure) + +2007-08-03 20:25 +0000 [r78103] Mark Michelson + + * main/config.c, channels/chan_sip.c, include/asterisk/config.h: + Changed the behavior of sip's realtime_peer function to match the + corresponding way of matching for non-realtime peers. Now matches + are made on both the IP address and port number, or if the + insecure setting is set to "port" then just match on the IP + address. In order to accomplish this, I also added a new API + call, ast_category_root, which returns the first variable of an + ast_category struct + +2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant + + * apps/app_voicemail.c: (closes issue #10194) Reported by: + blitzrage Patches: bug0010194 uploaded by vovochka Tested by: + blitzrage Fix a problem when you call Voicemail() with multiple + mailboxes specified and ODBC_STORAGE is in use. The audio part of + the message was only given to the first mailbox specified. + + * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some + improvements to lock debugging. These changes take effect with + DEBUG_THREADS enabled and provide the following: * This will keep + track of which locks are held by which thread as well as which + lock a thread is waiting for in a thread-local data structure. A + reference to this structure is available on the stack in the + dummy_start() function, which is the common entry point for all + threads. This information can be easily retrieved using gdb if + you switch to the dummy_start() stack frame of any thread and + print the contents of the lock_info variable. * All of the + thread-local structures for keeping track of this lock + information are also stored in a list so that the information can + be dumped to the CLI using the "core show locks" CLI command. + This introduces a little bit of a performance hit as it requires + additional underlying locking operations inside of every + lock/unlock on an ast_mutex. However, the benefits of having this + information available at the CLI is huge, especially considering + this is only done in DEBUG_THREADS mode. It means that in most + cases where we debug deadlocks, we no longer have to request + access to the machine to analyze the contents of ast_mutex_t + structures. We can now just ask them to get the output of "core + show locks", which gives us all of the information we needed in + most cases. I also had to make some additional changes to astmm.c + to make this work when both MALLOC_DEBUG and DEBUG_THREADS are + enabled. I disabled tracking of one of the locks in astmm.c + because it gets used inside the replacement memory allocation + routines, and the lock tracking code allocates memory. This + caused infinite recursion. + + * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control + frames when the mohinterpret option is set to "passthrough". This + was pointed out by Kevin in the middle of a training session. + + * channels/chan_iax2.c: Don't reuse the timespec that was set to 0 + in the previous timedwait as it will just return immediately. + Also, fix some logic so the thread's lock isn't unlocked twice in + the weird case of dynamic threads getting acquired right after a + timeout. (pointed out by SteveK) + +2007-08-02 21:53 +0000 [r77993-77996] Jason Parker + + * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we + actually allow 6 chars to be sent. Also make note of the "A" + option of date format. Issue 9779, modifications by DEA, wedhorn, + and myself. + + * channels/chan_skinny.c: If a device disconnects, the session will + go away. If this happens during call setup, we need to give up. + Issue 10325. + +2007-08-02 19:25 +0000 [r77949] Russell Bryant + + * channels/chan_iax2.c: Fix the case where a dynamic thread times + out waiting for something to do during the first time it runs. + This shouldn't ever happen, but we should account for it anyway. + (pointed out by pete, who works with mihai) + +2007-08-02 18:42 +0000 [r77947] Jason Parker + + * channels/chan_skinny.c: Make sure we clear the prompt status + message on a hangup. Also rearrange messages to better fit with + what a wireshark trace shows it should be. Issue 10299, initial + patch and solution by sbisker, modified by me to fit with + wireshark trace. + +2007-08-02 18:21 +0000 [r77945] Steve Murphy + + * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 + line This patch hopefully solves 10141; The user is running with + it, and it doesn't appear to harm asterisk's operation, and may + prevent a crash. I'll store it in 1.2, as we have shut down + support on 1.2, but since I developed the patch before support + finished, and it might affect 1.4 and trunk, I'm going ahead with + it. ........ + +2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant + + * channels/chan_iax2.c: Fix another race condition in the handling + of dynamic threads. If the dynamic thread timed out waiting for + something to do, but was acquired to perform an action + immediately afterwords, then wait on the condition again to give + the other thread a chance to finish setting up the data for what + action this thread should perform. Otherwise, if it immediately + continues, it will perform the wrong action. (reported on IRC by + mihai, patch by me) (related to issue #10289) + + * channels/chan_iax2.c: Add another sanity check to + vnak_retransmit(). This check ensures that frames that have + already been marked for deletion don't get retransmitted. (closes + issue #10361, patch from mihai) + +2007-08-02 15:15 +0000 [r77890-77894] Jason Parker + + * channels/chan_skinny.c: Make sure that we show the correct + extension if dialed from a macro "From: 5555" rather than "From: + s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, + tested by sbisker + + * channels/chan_skinny.c: Put in some additional debug information + for softkey/stimulus messages. Issue 10291, patch by DEA. + +2007-08-01 22:16 +0000 [r77887] Russell Bryant + + * channels/chan_iax2.c: Fix some race conditions which have been + causing weird problems in chan_iax2. The most notable problem is + that people have been seeing storms of VNAK frames being sent due + to really old frames mysteriously being in the retransmission + queue and never getting removed. It was possible that a dynamic + thread got created, but did not acquire its lock before the + thread that created it signals it to perform an action. When this + happens, the thread will sleep until it hits a timeout, and then + get destroyed. So, the action never gets performed and in some + cases, means a frame doesn't get transmitted and never gets freed + since the scheduler never gets a chance to reschedule + transmission. Another less severe race condition is in the + handling of a timeout for a dynamic thread. It was possible for + it to be acquired to perform at action at the same time that it + hit a timeout. When this occurs, whatever action it was acquired + for would never get performed. (patch contributed by Mihai and + SteveK) (closes issue #10289) (closes issue #10248) (closes issue + #10232) (possibly related to issue #10359) + +2007-08-01 22:14 +0000 [r77886] Tilghman Lesher + + * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does + not compile cleanly (missing def) + +2007-08-01 21:08 +0000 [r77883] Jason Parker + + * channels/chan_skinny.c: Fix an issue that caused one-way audio on + some newer devices (specifically the 7921), due to sending + packets in the wrong order during hangup. Also make sure we clear + tones/messages on the correct line/instance. Issue 10291, patch + by DEA, tested by sbisker and myself. + +2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp + + * main/cli.c: (closes issue #10351) Reported by: ftarz Some + platforms don't like it when you pass NULL to vsnprintf so pass + "" instead. + + * include/asterisk/threadstorage.h, channels/chan_mgcp.c, + apps/app_voicemail.c, main/acl.c, utils/smsq.c, + channels/chan_iax2.c: Add some fixes for building on Solaris. + + * main/utils.c: Whoops, I meant R_5 not R5. + + * configure, configure.ac: And for my last trick... make sure that + if gethostbyname_r is exported by a library that it is used. + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/utils.c: Extend autoconf logic to determine which version of + gethostbyname_r is on the system. + +2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson + + * apps/app_queue.c: Fixes an issue I introduced to queues wherein a + queue with joinempty=yes would kick people out of the queue + because of erroneously thinking the 'n' option was in use. + (closes issue #10320, reported by jfitzgibbon, patched by me, + tested by blitzrage and me) Thank you blitzrage for all the + testing you've done lately with queues! It's much appreciated! + + * apps/app_queue.c: If a queue uses dynamic realtime members, then + the member list should be updated after each attempt to call the + queue. This fixes an issue where if a caller calls into a queue + where no one is logged in, they would wait forever even if a + member logged in at some point. (closes issue #10346, reported by + and tested by blitzrage, patched by me) + +2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon + + * apps/app_rpt.c: Much newer version, 0.70 with much additions + + * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF + receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in + logic in TONE_VERIFY/RELAX mode in chan_zap. + +2007-07-31 20:59 +0000 [r77844] Steve Murphy + + * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 + line This probably isn't super-general, but it's a first stab at + using kill -11 to generate a core file instead of gcore. ........ + +2007-07-31 16:17 +0000 [r77831] Joshua Colp + + * res/res_speech.c, include/asterisk/speech.h: Add a flag to the + speech API that allows an engine to set whether it received + results or not. + +2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming + + * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without + DEBUG_THREADS or it does nothing + +2007-07-31 15:21 +0000 [r77824] Mark Michelson + + * channels/chan_sip.c: This patch makes Asterisk send 100 Trying + provisional responses upon receipt of re-invites. This makes it + so that if there are two or more Asterisk servers between + endpoints, the Asterisk servers will not keep retransmitting the + re-invites. (closes issue #10274, reported by cstadlmann, patched + by me with approval from file) + +2007-07-30 20:17 +0000 [r77795] Jason Parker + + * main/say.c: Applications like SayAlpha() should not hang up the + channel if you request an "unknown" character such as a comma. + Instead, skip the character and move on. Issue 10083, initial + patch by jsmith, modified by me. + +2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant + + * channels/chan_iax2.c: Fix an issue that could potentially cause + corruption of the global iax frame queue. In the network_thread() + loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE + macro. However, to remove an element of the list within this + loop, it used AST_LIST_REMOVE, instead of + AST_LIST_REMOVE_CURRENT, which I believe could leave some of the + internal variables of the SAFE macro invalid. Mihai says that he + already made this change in his local copy and it didn't help his + VNAK storm issues, but I still think it's wrong. :) + + * res/res_agi.c: (closes issue #10279) Reported by: seanbright + Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by + seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch + uploaded by seanbright (license 71) Allow the "agi_network: yes" + line to be printed out in the AGI debug output. Also, allow + partial writes to be handled when writing out this line just like + it is for all of the others. + + * main/channel.c: file and I both committed changes for issue + #10301. Remove a duplicated assignment to restore the original + value of the previous channel. + +2007-07-30 18:43 +0000 [r77783] Tilghman Lesher + + * /, res/res_agi.c: Merged revisions 77782 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) + | 2 lines Revert change in revision 71656, even though it fixed a + bug, because many people were depending upon the (broken) + behavior. ........ + +2007-07-30 17:29 +0000 [r77780] Russell Bryant + + * main/channel.c: (closes issue #10301) Reported by: fnordian + Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian + (license 110) Additional changes by me Fix some problems in + channel_find_locked() which can cause an infinite loop. The + reference to the previous channel is set to NULL in some cases. + These changes ensure that the reference to the previous channel + gets restored before needing it again. I'm not convinced that the + code that is setting it to NULL is really the right thing to do. + However, I am making these changes to fix the obvious problem and + just leaving an XXX comment that it needs a better explanation + that what is there now. + +2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp + + * res/res_features.c: (closes issue #10327) Reported by: kkiely + Instead of directly mucking with the extension/context/priority + of the channel we are transferring when it has a PBX simply call + ast_async_goto on it. This will ensure that the channel gets + handled properly and sent to the right place. + + * main/channel.c: (closes issue #10301) Reported by: fnordian + Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian + (license 110) Restore previous behavior where if we failed to + lock the channel we wanted we would return to exactly the same + point as if we had just reentered the function. + + * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 + lines (closes issue #10334) Reported by: ramonpeek Pass through + the return value from macro_exec through the MacroIf application. + ........ + +2007-07-27 18:15 +0000 [r77571] Tilghman Lesher + + * res/res_odbc.c: Missing newline + +2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp + + * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois + Patches: cdr_pgsql.patch uploaded by prashant (license 114) + Finish the Postgresql connection after the log messages are + printed so we don't access invalid memory. + + * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm + Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by + julianjm (license 99) Clear ONHOLD flag when decrementing the + onHold peer count. If we did not do this the count may keep + decreasing. + +2007-07-27 14:30 +0000 [r77490] Mark Michelson + + * channels/chan_sip.c: "re-invite" was misspelled + +2007-07-26 23:19 +0000 [r77460] Joshua Colp + + * main/channel.c: (closes issue #10302) Reported by: litnialex If a + DTMF end frame comes from a channel without a begin and it is + going to a technology that only accepts end frames (aka INFO) + then use the minimum DTMF duration if one is not in the frame + already. + +2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming + + * doc/mp3.txt: change protocol for downloads as well + + * doc/mp3.txt, sounds/Makefile: use new canonical name for download + server + +2007-07-26 21:23 +0000 [r77410] Russell Bryant + + * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined + in trunk, but not in 1.4. When Asterisk is compiled under dev + mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to + define it in the same way that trunk does. Also, revert the + change that added this define in the Makefile The advantage to + doing it this way is that buildopts.h gets installed when you + install Asterisk. Then, when building any out of tree modules, or + building asterisk-addons, these modules know which options the + rest of Asterisk was built with. + +2007-07-26 20:35 +0000 [r77380] Mark Michelson + + * Makefile, main/logger.c: Fixes to get ast_backtrace working + properly. The AST_DEVMODE macro was never defined so the majority + of ast_backtrace never attempted compilation. The makefile now + defines AST_DEVMODE if configure was run with --enable-dev-mode. + Also, changes were made to acccomodate 64 bit systems in + ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for + their roles in allowing me to get this committed + +2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher + + * main/logger.c: Missed one + + * main/logger.c: Oops, that builtin define should be all-lowercase. + +2007-07-26 18:30 +0000 [r77318] Mark Michelson + + * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur, + meaning free gets called on the same variable twice. This patch + sets the connection to NULL after calls to PQfinish so that the + problem does not occur. Also in this patch, prashant_jois + informed me that it is safe to pass a null pointer to PQfinish, + so I have removed the check for conn's existence from + my_unload_module. (closes issue 10295, reported by junky, patched + by me with input from prashant_jois) + +2007-07-25 22:39 +0000 [r77191] Steve Murphy + + * apps/app_meetme.c: This fix solves problem with intense squelch + noise when someone joins conf in bug 9430; We repro'd the problem + with meetme opts of 'CciMo'; Josh Colp supplied this patch, and + I'm applying it. It looks like playing the recorded username will + louse up the next thing played into the channel. Josh rearranged + the code so as to start things over before playing data directly + into the conference. + +2007-07-25 22:16 +0000 [r77176] Joshua Colp + + * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd + Add SPEECH_DTMF_TERMINATOR variable so the user can specify the + digit to terminate a DTMF string with. If none is specified then + no terminator will be used. + +2007-07-25 21:52 +0000 [r77154] Mark Michelson + + * main/channel.c: chan->emulate_dtmf_duration is an unsigned int, + not a signed int, so use %u instead of %d in the format string + +2007-07-25 20:23 +0000 [r77116-77136] Jason Parker + + * /: so are my fingers... + + * /: autotagexternals script is still obviously misbehaving... + + * /: use autotagged externals + +2007-07-25 17:14 +0000 [r77071] Joshua Colp + + * configure, acinclude.m4: Fix autoconf logic for finding OpenH323 + when it is not in the first place searched (/usr/share/openh323). + +2007-07-25 09:34 +0000 [r77022] Luigi Rizzo + + * main/rtp.c: set the sequence number in a frame for all frame + types + +2007-07-25 00:18 +0000 [r76983] Steve Murphy + + * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 + line this fixes bug 10293, where the error message because + defaultzone or loadzone was not defined was confusing ........ + +2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher + + * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 + Jul 2007) | 2 lines Oops, res contains the error code, not errno. + I was wondering why a mutex was reporting "No such file or + directory"... ........ + + * main/app.c: Found another place where we should be using the + umask (thanks jcmoore) + +2007-07-24 Jason Parker + + * Asterisk 1.4.9 released. + +2007-07-24 16:42 +0000 [r76803-76805] Jason Parker + + * channels/chan_iax2.c: Don't create the Asterisk channel until we + are starting the PBX on it. (ASA-2007-018) + +2007-07-24 16:26 +0000 [r76801] Mark Michelson + + * apps/app_queue.c: Added a membercount variable to call_queue + struct which keeps track of the number of logged in members in a + particular queue. This makes it so that the 'n' option for + Queue() can act properly depending on which strategy is used. If + the strategy is roundrobin, rrmemory, or ringall, we want to ring + each phone once before moving on in the dialplan. However, if any + other strategy is used, we will only ring one phone since it + cannot be guaranteed that a different phone will ring on + subsequent attempts to ring a phone. As a side effect of this, + the QUEUE_MEMBER_COUNT dialplan function now just reads the + membercount variable instead of traversing through the member + list to figure out how many members there are. Special thanks to + blitzrage for helping to test this out. (closes issue #10127, + reported by bcnit, patched by me, tested by blitzrage) + +2007-07-23 22:38 +0000 [r76708] Tilghman Lesher + + * apps/app_voicemail.c: It was our stated intention for 1.4 that + files created in app_voicemail should depend upon the umask. + Unfortunately, mkstemp() creates files with mode 0600, regardless + of the umask. This corrects that deficiency. + +2007-07-23 18:59 +0000 [r76656] Jason Parker + + * channels/chan_skinny.c: Fix some incorrect softkey labels in + messages. Don't try to play dialtone in some unimplemented + features. + +2007-07-23 18:29 +0000 [r76654] Joshua Colp + + * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul + 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not + force channel format changes when a generator is present. The + generator may have changed the formats itself and changing them + back would cause issues. ........ + +2007-07-23 17:57 +0000 [r76620] Jason Parker + + * channels/chan_skinny.c: Don't try to queue up hold/unhold frames + on a non-existent channel. Issue 10276. + +2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp + + * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux + by putting the value of the digit char in an int. + + * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 + lines (closes issue #10236) Reported by: homesick Patches: + rpid_1.4_75840.patch uploaded by homesick (license 91) Accept + Remote Party ID on guest calls. ........ + + * channels/chan_skinny.c: (closes issue #10268) Reported by: + mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak + (license 7) Add another OS that has to use the Macros for byte + ordering. + +2007-07-23 12:25 +0000 [r76485] Russell Bryant + + * channels/chan_iax2.c: Use a signed integer for storing the number + of bytes in the packet read from the network. Using an unsigned + value here made it impossible to handle an error returned from + recvfrom(). Furthermore, in the case that recvfrom() did return + an error, this would cause a crash due to a heap overflow. + (closes issue #10265, reported by and fix suggested by + timrobbins) + +2007-07-21 02:02 +0000 [r76227] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | + 4 lines Backport a fix for a memory leak that was fixed in trunk + in reivision 76221 by rizzo. The memory used for the localaddr + list was not freed during a configuration reload. ........ + +2007-07-20 21:36 +0000 [r76211] Steve Murphy + + * sounds/Makefile: This patch from 10249 is worth applying! It + prevents downloading sound files if they are already downloaded. + Darn Practical, if you ask me + +2007-07-20 21:03 +0000 [r76174-76178] Jason Parker + + * channels/chan_skinny.c: Allow getting a call from an existing + "sub" channel. Cancel ringing if endpoint hangs up before + answering. Fixes were backported from trunk (there was apparently + a bit of confusion during merge of a previous patch). (closes + issue #10241) + + * main/manager.c: Eliminate a compiler warning with gcc 4.2 by + constifying a char * + + * channels/chan_skinny.c: It's possible for sub->owner to be NULL + here if you cancel the call immediately after/during sending a + digit. + +2007-07-20 18:42 +0000 [r76139] Mark Michelson + + * apps/app_directory.c: When using users.conf for the entries in + the directory, if multiple users had the same last name, only the + first user listed would be available in the directory. (closes + issue #10200, reported by mrskippy, patched by me) + +2007-07-20 18:22 +0000 [r76132] Russell Bryant + + * main/channel.c: Use the define that specifies the default length + of an artificially created DTMF digit in the ast_senddigit() + function. The define is set to 100ms by default, which is the + same thing that this function was using. But, using the define + lets changes take effect in this case, as well as the others + where it was already used. + +2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 + lines (closes issue #10247) Reported by: fkasumovic Patches: + chan_sip.patch uploaded by fkasumovic (license #101) Drop any + peer realm authentication entries when reloading so multiple + entries do not get added to the peer. ........ + + * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic + Patches: res_conver.patch uploaded by fkasumovic (license #101) + Use the last occurance of . to find the extension, not the first + occurance. + + * apps/app_queue.c: Move makeannouncement variable declaration to + proper place. + +2007-07-19 20:36 +0000 [r75980] Jason Parker + + * channels/chan_skinny.c: Remove some duplicate code. + +2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson + + * apps/app_queue.c: The diff on this looks pretty big but all I did + was remove a pointless if statement (always evaluates true). + + * apps/app_queue.c: Changes in handling return values of several + functions in app_queue. This all started as a fix for issue + #10008 but now includes all of the following changes: 1. + Simplifying the code to handle positive return values from ast + API calls. 2. Removing the background_file function. 3. The fix + for issue #10008 (closes issue #10008, reported and patched by + dimas) + +2007-07-19 15:53 +0000 [r75928] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | + 6 lines When processing full frames, take sequence number + wraparound into account when deciding whether or not we need to + request retransmissions by sending a VNAK. This code could cause + VNAKs to be sent erroneously in some cases, and to not be sent in + other cases when it should have been. (closes issue #10237, + reported and patched by mihai) ........ + +2007-07-18 22:59 +0000 [r75807] Jason Parker + + * channels/chan_skinny.c: Need to make sure we set milliseconds and + timestamp - pointed out by the recent ast_ time stuff from + Tilghman + +2007-07-18 21:09 +0000 [r75759] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | + 5 lines When traversing the queue of frames for possible + retransmission after receiving a VNAK, handle sequence number + wraparound so that all frames that should be retransmitted + actually do get retransmitted. (issue #10227, reported and + patched by mihai) ........ + +2007-07-18 20:40 +0000 [r75749] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007) + | 2 lines Store prior to copy (closes issue #10193) ........ + +2007-07-18 20:17 +0000 [r75732] Jason Parker + + * channels/chan_skinny.c: Umm, why are we transmitting dialtone on + cfwdall? + +2007-07-18 20:00 +0000 [r75712] Joshua Colp + + * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c, + pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these + Asterisk won't build under devmode using GCC 4.2. + +2007-07-18 19:54 +0000 [r75707-75711] Jason Parker + + * channels/chan_skinny.c: Fixes for 7935/7936 conference phones. + Issue 9245, patch by slimey. + + * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue + 9887, patches by DEA + +2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard + + * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007) + | 1 line removed the word 'pissed' from ast_log(...) function + call for BE-90 ........ + +2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp + + * channels/chan_sip.c: Few more places that needs to check for + onhold state. + + * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar + It is possible for hold status to exist without call limits set, + so we need to ensure update_call_counter is executed regardless. + + * channels/chan_h323.c: Don't bother reloading chan_h323 if it did + not load successfully in the first place. This would otherwise + cause a crash. + + * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record + the threadid of each running thread before shutting them down as + the thread themselves may change the value. + +2007-07-18 12:29 +0000 [r75529] Tilghman Lesher + + * apps/app_meetme.c: Using a freed frame causes crashes (closes + issue #9317) + +2007-07-17 Russell Bryant + + * Asterisk 1.4.8 released. + +2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant + + * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 + Jul 2007) | 3 lines Properly check for the length in the skinny + packet to prevent an invalid memcpy. (ASA-2007-016) ........ + + * main/rtp.c: cast arguments to ast_log so that it builds without + warnings for me + + * channels/iax2-parser.c, channels/iax2-parser.h, /, + channels/chan_iax2.c: Merged revisions 75444 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | + 5 lines Ensure that when encoding the contents of an ast_frame + into an iax_frame, that the size of the destination buffer is + known in the iax_frame so that code won't write past the end of + the allocated buffer when sending outgoing frames. (ASA-2007-014) + ........ + + * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | + 4 lines After parsing information elements in IAX frames, set the + data length to zero, so that code later on does not think it has + data to copy. (ASA-2007-015) ........ + +2007-07-17 20:40 +0000 [r75439] Joshua Colp + + * main/rtp.c: Ensure that the pointer to STUN data does not go to + unaccessible memory. (ASA-2007-017) + +2007-07-17 20:33 +0000 [r75437] Russell Bryant + + * res/res_agi.c: (issue #10210) Reported by: juggie Patches: + 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by: + juggie, blitzrage Log a warning if someone uses DeadAGI on a live + channel. + +2007-07-17 20:03 +0000 [r75405] Mark Michelson + + * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists + can return -1 on failure, so I need to be sure that we only enter + the if statement if it is successful. Related to my fix to issue + #10186 + +2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant + + * main/pbx.c: (closes issue #10209) Reported by: juggie Patches: + 10209-trunk-2.patch uploaded by juggie Tested by: juggie, + blitzrage In ast_pbx_run(), mark a channel as hung up after an + application returned -1, or when it runs out of extensions to + execute. This is so that code can detect that this channel has + been hung up for things like making sure DeadAGI is used on + actual dead channels, and is beneficial for other things, like + making sure someone doesn't try to start spying on a channel that + is about to go away. + + * res/res_agi.c: Remove a duplicated newline character in AGI debug + output. (closes issue #10207, patch by seanbright) + +2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming + + * main/dns.c, /: Merged revisions 75304 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007) + | 3 lines provide proper copyright/license attribution for this + structure that was copied from a BSD-licensed header file long, + long ago... ........ + + * /: another fix that is not needed here (finishing up 75251) + +2007-07-16 18:16 +0000 [r75253] Mark Michelson + + * apps/app_dial.c: Restoring functionality from 1.2 wherein + Retrydial will not exit if there is no announce file specified. + This change makes it so that if there is no announce file + specified, the application will continue until finished (or + caller hangs up). If a bogus announce file is specified, then a + warning message will be printed saying that the file could not be + found, but execution will still continue. (closes issue #10186, + reported by jon, patched by me) + +2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming + + * /: block change that is not relevant here + +2007-07-13 20:36 +0000 [r75108] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 + Jul 2007) | 3 lines Fix a couple potential minor memory leaks. + load_moh_classes() could return without destroying the loaded + configuration. ........ + +2007-07-13 20:15 +0000 [r75078] Mark Michelson + + * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul + 2007) | 5 lines Fixed an issue where chanspy flags were + uninitialized if no options were passed. What triggered this + investigation was an IRC chat where some people's quiet flags + were set while others' weren't even though none of them had + specified the q option. ........ + +2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 + Jul 2007) | 6 lines Ensure that adding a user to the list of + users of a specific music on hold class is not done at the same + time as any of the other operations on this list to prevent list + corruption. Using the global moh_data lock for this is not ideal, + but it is what is used to protect these lists everywhere else in + the module, and I am only changing what is necessary to fix the + bug. ........ + + * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | + 12 lines (closes issue #9660) Reported by: mmacvicar Patches + submitted by: bbryant, russell Tested by: mmacvicar, marco, + arcivanov, jmhunter, explidous When using a TDM400P (and probably + other analog cards) there was a chance that you could hang up and + pick the phone back up where it has been long enough to be not + considered a flash hook, but too soon such that the device + reports that it is busy and the person on the phone will only + hear silence. This patch makes chan_zap more tolerant of this and + gives the device a couple of seconds to succeed so the person on + the phone happily gets their dialtone. ........ + +2007-07-12 23:00 +0000 [r74998] Mark Michelson + + * channels/chan_agent.c: Change to my previous fix regarding agent + logoff soft. Now uses deferlogoff instead of loginstart since + loginstart is used after logoff. Thanks to makoto for pointing + this out and suggesting the fix. (closes issue #10178, reported + and patched by makoto, with modification by me) + +2007-07-12 20:42 +0000 [r74955] Steve Murphy + + * channels/chan_sip.c: This patch resolves 10143; thanks to irroot + for the patch; looked acceptable. Let the community decide if it + messes things up + +2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp + + * main/channel.c: Whoops... didn't want this to be returned to 0 + each iteration. + + * main/channel.c: When waiting for a digit ensure that a begin + frame was received with it, not just an end frame. (issue #10084 + reported by rushowr) + +2007-07-12 16:53 +0000 [r74839-74866] Jason Parker + + * channels/chan_skinny.c: It helps if I actually add this stuff for + the 7921 too - otherwise it won't actually do much of anything. + + * channels/chan_skinny.c: Add device ID for 7921 wireless skinny + phone + + * channels/chan_skinny.c: Fix dialing in skinny that was broken in + some cases. Issue 10136, fix provided by DEA. + +2007-07-12 15:53 +0000 [r74815] Joshua Colp + + * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul + 2007) | 2 lines Only print out a warning for situations where it + is actually helpful. (issue #10187 reported by denke) ........ + +2007-07-11 22:57 +0000 [r74767] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | + 5 lines The function make_trunk() can fail and return -1 instead + of a valid new call number. Fix the uses of this function to + handle this instead of treating it as the new call number. This + would cause a deadlock and memory corruption. (possible cause of + issue #9614 and others, patch by me) ........ + +2007-07-11 21:14 +0000 [r74722] Mark Michelson + + * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 + Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" + did not work...at all. Now it does. (closes issue #10178, + reported and patched by makoto, with slight modification for 1.4 + and trunk by me) ........ + +2007-07-11 18:34 +0000 [r74657] Russell Bryant + + * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 + Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows + the condition that uses LIKE. This fixes realtime extensions with + ODBC. (closes issue #10175, reported by stuarth, patch by me) + ........ + +2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy + + * Makefile: This fixes 10172, where the entire man8 dir gets + removed during an uninstall of asterisk + + * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt: + further reversion of previously applied floating point stuff for + expr2 + +2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp + + * channels/chan_phone.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Instead of + figuring out kernel versions that have compiler.h and not... + let's just use autoconf to check for it's presence. (issue #10174 + reported by francesco_r) + + * channels/chan_phone.c: Only check if we need to do a SIGMA based + tone generation if we have a card. (issue #10179 reported by + mikowhy) + +2007-07-10 23:32 +0000 [r74476] Mark Michelson + + * apps/app_voicemail.c: Forwarding a message with IMAP storage was + storing the message in the sender's box instead of the forwarded + mailbox. (closes issue #10138, reported and patched by jaroth) + +2007-07-10 19:58 +0000 [r74374-74428] Jason Parker + + * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 + lines Fix an issue where it was possible to have a service level + of over 100% Between the time recalc_holdtime and update_queue + was called, it was possible that the call could have been hungup. + Move both additions to the same place, so this won't happen. + Issue 10158, initial patch by makoto, modified by me. ........ + + * main/dns.c: Don't use #if to check if something is defined - use + #ifdef instead. Pointed out by kpfleming + + * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul + 2007) | 4 lines Fix an issue with wrapuptime not working when + using AgentLogin. Issue 10169, patch by makoto, with a minor mod + by me to not re-break issue 9618 ........ + + * main/dns.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Merged revisions 74373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 + lines Use res_ndestroy on systems that have it. Otherwise, use + res_nclose. This prevents a memleak on NetBSD - and possibly + others. Issue 10133, patch by me, reported and tested by scw + ........ + +2007-07-10 Russell Bryant + + * Asterisk 1.4.7.1 released. + +2007-07-10 16:00 +0000 [r74323] Russell Bryant + + * res/res_musiconhold.c: fix an uninitialized variable + +2007-07-10 15:38 +0000 [r74317] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 + lines Fix a small typo in description in of Voicemail() + application. Issue 10170, patch by casper. ........ + +2007-07-10 15:31 +0000 [r74314] Russell Bryant + + * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 + Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue + #10075, this part reported by jmls on IRC, patch by me) ........ + +2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp + + * /, main/app.c: Merged revisions 74264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 + lines Ensure the group information category exists before trying + to do a string comparison with it. (issue #10171 reported by + mlegas) ........ + + * channels/chan_sip.c: Only spit out an inringing warning message + when it is applicable. Since call limits are already toast in + realtime let's not scare the user if they are using it. (issue + #10166 reported by bcnit) + +2007-07-09 Russell Bryant + + * Asterisk 1.4.7 released. + +2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant + + * configure, configure.ac: Update the configure script to check for + a required function that is not present in the 1.2 version of + libpri. This will prevent the configure script from thinking that + it has compatible libpri support for Asterisk 1.4, when it + actually does not because the installed version is from 1.2. + + * res/res_musiconhold.c: (closes issue #10123) Reported by: + blitzrage Patches submitted by: juggie, qwell, me Tested by: + blitzrage When trying to find a music on hold class to use, try + all of the options, instead of only the first one that is set. + Also, change the MusicOnHold applications to not hang up on the + channel when a class can not be found. + +2007-07-09 20:19 +0000 [r74159] Jason Parker + + * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 + lines Several chan_zap options were not working on reload because + they were arbitrarily disallowed when reloading some/most PRI + options (such as signalling) was disallowed. Options such as + polarityonanswerdelay and answeronpolarityswitch can safely be + changed on a reload. This corrects that behavior. Issue 9186, + patch by tzafrir. ........ + +2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson + + * apps/app_queue.c: Forgot to get rid of an extraneous debug + message. + + * apps/app_queue.c: The n option for Queue should make the queue + exit immediately after failure to reach any members and should + not be dependent on the timeout value passed to Queue (closes + issue #10127, reported by bcnit, repaired by me) + +2007-07-09 15:32 +0000 [r74082] Joshua Colp + + * channels/chan_skinny.c: Only destroy the scheduler context if it + was allocated. (issue #10124 reported by gzero) + +2007-07-09 14:57 +0000 [r74047] Mark Michelson + + * apps/app_voicemail.c: Fixed a logic error in leave_voicemail. + Pass the mailbox instead of the context to inbox_count when the + context is "default." (closes issue #10135, reported by yannj, + repaired by me) + +2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp + + * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread + synchronization tweaks. (issue #10124 reported by gzero) + + * configure, acinclude.m4: Use AC_CHECK_HEADER to check for + ptlib/openh323 to allow for cross compiling. (issue #9675 + reported by zandbelt) + +2007-07-09 04:03 +0000 [r73985] Tilghman Lesher + + * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while + 'make progdocs'. (Closes issue #10104) + +2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp + + * main/cdr.c: Give Agent channel names priority when doing CDR + merging. (issue #10011 reported by krtorio) + + * pbx/pbx_config.c: Add a few sanity checks when writing out the + dialplan. (issue #10157 reported by dome) + +2007-07-08 09:47 +0000 [r73849] Olle Johansson + + * channels/chan_sip.c: While tracking down a bug, I need some more + history. Dumphistory is very useful, indeed. + +2007-07-06 23:02 +0000 [r73769] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | + 4 lines If a sip_pvt struct has already registered an extension + state callback, remove the old one before adding a new one. If + this isn't done, Asterisk will crash. (issue #10120) ........ + +2007-07-06 16:36 +0000 [r73727] Mark Michelson + + * apps/app_voicemail.c: Fixing a rare case which causes voicemail + to crash when compiled with IMAP storage. inboxcount has the + possibility of finding an "interactive" vm_state when no + persistent "non-interactive" vm_state exists for that mailbox. If + this should happen when someone attempts to leave a message, it + results in a crash. This patch, along with my commit in revision + 72670 fix issue 10053, reported by jaroth. closes issue #10053 + +2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant + + * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 + Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras + Patches submitted by: Corydon76 Tested by: apsaras Fix a problem + with MSSQL 2005 by explicitly stating that '\' is being used as + an escape character. ........ + + * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | + 7 lines (closes issue #10125) Reported by: makoto Patches + submitted by: makoto This fixes a crash in chan_sip that happens + when the bindaddr setting is not valid on Asterisk startup, gets + fixed, and then a reload gets issued. ........ + +2007-07-06 15:27 +0000 [r73675] Mark Michelson + + * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 + Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. + (issue 9618, reported by jiddings, patched by moi) closes issue + #9618 ........ + +2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant + + * BUGS: fix a little spelling error + + * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop + the monitor thread if it was never started. (closes issue #10124, + reported by gzero, fixed by me) + + * channels/chan_iax2.c: copy from the correct buffer when deferring + a full frame (related to issue #9937) + + * channels/chan_iax2.c: * Store the call number that a thread is + processing without the full frame bit set to ease debugging * + When deferring a full frame for processing, stick it into the + queue for the thread that is processing frames for that call, not + the one that read the current frame and is about to go back into + the idle list (related to issue #9937) + +2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming + + * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) + | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just + like we don't support it for G.729 ........ + +2007-07-05 20:50 +0000 [r73512] Russell Bryant + + * res/res_features.c: Pass HOLD and UNHOLD frames to the other + channel when they are returned from a native bridge function. + This fixes a problem where when two zap channels are natively + bridged and one does a flash hook, the other channel did not + receive music on hold. (Reported to me directly by Doug Bailey at + Digium) + +2007-07-05 19:18 +0000 [r73467] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 + lines Copy language information to the dialog structure when + calling a peer for situations where a PBX may be started on the + dialed channel. (issue #10121 reported by clegall_proformatique) + ........ + +2007-07-05 15:59 +0000 [r73400] Mark Michelson + + * apps/app_queue.c: Correcting a minor CLI bug I found. When + issuing the queue show command, if you type queue show and then + press tab, you can continue pressing tab and it will keep + auto-completing queue names even though only 1 queue can be used + as an argument. + +2007-07-05 15:28 +0000 [r73398] Russell Bryant + + * channels/chan_vpb.cc, channels/Makefile: Make this module build + for me in dev-mode + +2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp + + * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 + lines Tweak spy locking. (issue #9951 reported by welles) + ........ + + * channels/chan_local.c, /: Merged revisions 73318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul + 2007) | 2 lines Actually check to make sure a PBX was started on + one of the Local channels instead of blindly assuming it was. + (issue #10112 reported by makoto) ........ + + * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 + lines Reset ServicelevelPerf variable back to 0 if we are unable + to calculate it each time... otherwise we will get previous + values. (issue #10117 reported by noriyuki) ........ + +2007-07-04 14:53 +0000 [r73208-73253] Christian Richter + + * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 + Jul 2007) | 1 line bchannel configurations like echocancel and + volume control, need to be setuped on inbound calls too. ........ + + * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 + Jul 2007) | 1 line bad bug in overlapdial case, we called + start_pbx multiple times, because the state wasn't changed.. + ........ + +2007-07-03 20:17 +0000 [r73143] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile, + main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing + expr floating patch from 1.4; too much of a behavior change. If + you want this fix, try trunk instead. bug 9508. + +2007-07-03 15:42 +0000 [r73104-73106] Jason Parker + + * /: What the heck. This should not have happened. + + * /: use autotagged externals + +2007-07-03 12:38 +0000 [r73053] Tilghman Lesher + + * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) + | 2 lines RetryDial should accept a 0 argument, but it does not, + because atoi does not distinguish between 0 and error (closes + issue #10106) ........ + +2007-07-03 08:17 +0000 [r73005] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 + Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only + be called from mISDN Source channels.. #9449 ........ + +2007-07-02 20:16 +0000 [r72933] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput, + main/Makefile, main/ast_expr2.h, main/ast_expr2.y, + main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support + for floating point numbers added to ast_expr2 $\[...\] exprs. + Fixes bug 9508, where the expr code fails with fp numbers. The + MATH function returns fp numbers by default, so this fix is + considered necessary. + +2007-07-02 18:18 +0000 [r72926] Russell Bryant + + * main/manager.c: Remove a bogus comment and add proper locking to + the handler function for the CLI command to show information on + manager actions. + +2007-07-02 14:32 +0000 [r72888] Joshua Colp + + * main/channel.c: Added additional DTMF debug messages for when + emulation occurs. + +2007-07-02 08:41 +0000 [r72850-72852] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 72585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | + 1 line check if the bchannel stack id is already used, if so + don't use it a second time. Also added a release_chan lock, so + that the same chan_list object cannot be freed twice. chan_misdn + does not crash anymore on heavy load with these changes. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 72099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | + 1 line simplified generation for dummy bchannels, also we mark + them as dummies, so they are not used later as real-bchannels, + optimized the RESTART mechanisms, we block a channel now on + cause:44, and send out a RESTART automatically, then on reception + of RESTART_ACKNOWLEDGE we unblock the channel again. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged + revisions 72087 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | + 1 line simplified channel finding and locking a lot. removed + unnecessary #ifdefed areas. ........ + +2007-07-01 23:52 +0000 [r72806] Russell Bryant + + * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | + 5 lines When appending lines to call files to keep track of + retries, write a leading newline just in case the original call + file did not have a newline at the end. This fix is in response + to a problem I saw reported on the asterisk-users mailing list. + ........ + +2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant + + * configure, configure.ac: Tweak the configure script so that error + output isn't spewed to the console when searching for GTK2 libs, + and they aren't found. + + * formats/format_pcm.c: give format_pcm a more concise destription + +2007-06-29 19:07 +0000 [r72665] Luigi Rizzo + + * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for + absence of the function. This was already done in trunk. + +2007-06-29 Russell Bryant + + * Asterisk 1.4.6 released. + +2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp + + * main/cdr.c: Minor change for older GCC versions. + + * Makefile, configure, configure.ac, makeopts.in: Backport fix for + GCC versions without support for declaration-after-statement. + +2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher + + * main/manager.c: Issue 10055 - Change memory allocation to use the + heap for a command, since the output has the potential to + overflow the stack (as it did here) + + * res/res_jabber.c: Fix 1.4 breakage + +2007-06-28 19:44 +0000 [r72493] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in: regenerate the + configure script for rizzo + +2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo + + * configure.ac: add a check for gethostbyname_r so we can simplify + the handling e.g. in utils.c Also add comments on a couple of + features which are not working on FreeBSD. All the above has been + already done in trunk so the merge must be blocked. Can someone + please regenerate ./configure ? + + * Makefile, channels/chan_zap.c, main/say.c: Add + -Wdeclaration-after-statement to AST_DEVMODE flags to catch + variable declarations in the middle of a block. Fix the few + instances of the above spotted out by the compiler. All of this + has been already done or is not applicable in trunk, so the merge + of this change will be blocked. + + * apps/app_meetme.c: cast a time_t so that it does not conflict + with the print format. This change was already done on trunk so + this change needs to be blocked from merging. + +2007-06-27 23:29 +0000 [r72383] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | + 3 lines Reinstating patch. This actually fixes the problem, + however I was running a development branch without it and + mistakenly thought it wasn't fixed. Fixes issue #10010, and + #9654: 100% CPU usage caused by an asterisk console losing it's + controlling terminal. ........ + +2007-06-27 23:25 +0000 [r72381] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun + 2007) | 2 lines Update documentation to clarify variable usage + with MixMonitor. (issue #9494 reported by netoguy) ........ + +2007-06-27 23:03 +0000 [r72335] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72333 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | + 2 lines Reverted changes for earlier revisions 72259 to 72261. + Issue #9654, #10010 ........ + +2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp + + * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to + our list. Since these are dynamic payloads the other side + shouldn't care. (issue #9426 reported by irroot) + + * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 + lines Fix issue where queue log events might be missing. (issue + #7765 reported by mtryfoss) ........ + +2007-06-27 21:08 +0000 [r72272] Russell Bryant + + * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | + 5 lines Fix a minor issue with parsing the priority number. You + could have as much whitespace as you want around a numeric + priority, but you couldn't have any whitespace around a special + priority like "n" or "hint". (issue #10039, reported by mitheloc, + fixed by me) ........ + +2007-06-27 20:46 +0000 [r72260] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72259 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | + 4 lines Fixes 100% load when controlling terminal disappears. + Issue #9654, #10010 ........ + +2007-06-27 20:25 +0000 [r72257] Joshua Colp + + * main/channel.c, /: Merged revisions 72256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 + lines I may possibly get shot for doing this... but... defer CDR + processing until after the channel has been dealt with. This + should eliminate all of the issues with channels going funky + (SIP/PRI) when you are posting CDRs to a database that is either + slow or unavailable and do not want to enable batching. ........ + +2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming + + * channels/chan_zap.c: use the proper type for storing group number + bits so that if someone specifies 'group=42' it will actually + work instead of being silently ignored + +2007-06-27 18:40 +0000 [r72182-72185] Jason Parker + + * apps/app_voicemail.c: Fix another problem in voicemail with + missing symbols. Issue 10074, patch by kryptolus, extended to + include #if 0'd blocks (just in case) + +2007-06-27 17:31 +0000 [r72148] Joshua Colp + + * main/channel.c: Make the ast_read_noaudio API call behave better + under circumstances where DTMF emulation was happening and a + generator was setup. (issue #10065 reported by stevefeinstein) + +2007-06-27 17:10 +0000 [r72125] Jason Parker + + * channels/chan_gtalk.c: Don't modify a variable that we don't want + modified. Make a copy of it instead. Issue 10029, patch by + phsultan with slight modifications by me (to remove needless + casts). + +2007-06-27 16:34 +0000 [r72112] Russell Bryant + + * main/rtp.c: Only output debug information related to RTCP + timestamps when RTCP debug is turned on (issue #10066, patch by + me) + +2007-06-27 07:58 +0000 [r72042] Christian Richter + + * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | + 1 line for inbound TE calls, we setup the bchannel when we get + the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. + removed some #if 0 areas which weren't used anymore. ........ + r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | + 1 line isdn_lib.c didn't compile ........ + +2007-06-27 00:58 +0000 [r72006] Joshua Colp + + * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. + +2007-06-26 23:02 +0000 [r71953] Mark Michelson + + * apps/app_voicemail.c: Removing a pointless line. This variable + was already set earlier and between then and this line, there is + no way that the values on the right side of the assignment could + have changed. + +2007-06-26 20:36 +0000 [r71915] Jason Parker + + * main/rtp.c: Don't dereference a pointer that may be NULL here. + Issue 10017. + +2007-06-26 19:00 +0000 [r71877] Mark Michelson + + * apps/app_voicemail.c: A few changes, the ultimate goal of which + is to keep better track of the number of messages that a mailbox + currently has. A description of the changes: 1. Changed the + "updated" field of the vm_state struct to act more as a binary + semaphore than a counting semaphore, since its current + implementation made the inboxcount function not work properly. + This change falls in line with a change made by UPenn with their + IMAP setup and helps to sync our changes with theirs. 2. + Eliminated some redundant calls to get_vm_state_by_mailbox inside + leave_voicemail 3. Use the play_folder variable to keep track of + the number of old and new messages in a mailbox as the messages + are deleted 4. Added an increment to the number of new messages + that was not there previously in the leave_voicemail function + +2007-06-26 15:47 +0000 [r71796] Mark Michelson + + * apps/app_voicemail.c: Fixing bug where the authuser was + mistakenly pulled from the mailbox string instead of the IMAP + user. (closes issue 10054, reported and patched by jaroth) + +2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) + | 2 lines Issue 10062 - Trying to move a message without + selecting one first results in memory corruption ........ + + * /, res/res_agi.c: Merged revisions 71656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) + | 2 lines Issue 10035 - handle_exec returns a result inconsistent + with all of the other AGI commands ........ + +2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp + + * channels/chan_h323.c: Build a peer as well when hash323 is + enabled in users.conf (issue #9599 reported by asagage) + + * channels/chan_agent.c: Minor tweak for queueing up the unhold + frame... this will teach me to do bugs while half asleep. (issue + #10046 reported by dimas) + +2007-06-25 12:40 +0000 [r71519] Russell Bryant + + * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue + #10048, Matti) + +2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 + lines Ignore other URIs after the first in a 300 Multiple Choice + response. (issue #10041 reported by homesick) ........ + + * main/cdr.c: Fix it so 1.4 actually compiles on my box. + + * channels/chan_agent.c: Check to make sure the channel pointer is + present before queueing up an unhold frame on it. (issue #10046 + reported by dimas) + +2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant + + * build_tools/prep_tarball: Include the menuselect-tree file in + tarballs to make builds from tarballs a little bit faster + + * main/asterisk.c, /: Merged revisions 71358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | + 2 lines Revert the patch from issue 9654 due to an unexpected + side effect ........ + +2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher + + * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so + peer->cdr is what we really wanted to use + + * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 + Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to + be able to set variables to the empty string. ........ + +2007-06-23 03:29 +0000 [r71230] Steve Murphy + + * main/cdr.c, res/res_features.c: This patch is meant to fix 8433; + where clid and src are lost via bridging. + +2007-06-22 22:44 +0000 [r71214] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 + Jun 2007) | 1 line fixed a bug that was introduced by copy and + paste in the last commit ..bchannels weren't cleaned properly. + ........ + +2007-06-22 15:38 +0000 [r71096-71123] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 70672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | + 1 line we activate the bchannels in TE mode on incoming calls + only when we want to connect the call. ........ + + * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 + Jun 2007) | 1 line forgot one place .. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 + Jun 2007) | 1 line on receiption of cause:44 we mark the channel + as in use and inform the user about the situation, we need to + test the RESTART stuff then. Also shuffled the + empty_chan_in_stack function after the bchannel cleaning + functions, to avoid race conditions. ........ + + * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 + Jun 2007) | 1 line when we send out a SETUP, but get no response, + we should cleanup everything after reception of a hangup. + ........ + + * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | + 1 line restart indicator 0x80 is correct, at least that's what + libpri does. ........ + + * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 + Jun 2007) | 1 line if the bridged partner is mISDN too we should + not send dtmf tones, they are transmitted inband always ........ + + * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 + Jun 2007) | 1 line if we have already some digits, we just stop + the tones. ........ + +2007-06-22 15:00 +0000 [r71068] Jason Parker + + * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged + revisions 71065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 + lines Fix a few silly usages of ast_playstream() - it only ever + returns 0... Issue 10035 ........ + +2007-06-22 14:53 +0000 [r71066] Brett Bryant + + * main/asterisk.c, /: Merged revisions 71064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | + 10 lines Fixed infinite loop when controlling terminal was lost + and return value of input function wasn't checked for errors. + This would cause 100% cpu to be taken up. (closes issue #9654, + issue #10010) Reported by: mnicholson, and eserra Idea for the + patch from mnicholson, patched by me ........ + +2007-06-22 14:10 +0000 [r71063] Steve Murphy + + * main/cdr.c: My conditions for merging amaflags info was naive; + DOCUMENTATION is the default, although null is possible; theft of + user-settable fields is not good. Just copy them, leave them + alone. + +2007-06-22 03:14 +0000 [r71003] Russell Bryant + + * channels/chan_iax2.c: Fix a small typo which ... well ... + completely broke chan_iax2. oops! (issue #9937, patch by me) + +2007-06-21 22:34 +0000 [r70949] Steve Murphy + + * main/cdr.c, /: Merged revisions 70948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 + line This little fix is in response to bug 10016, but may not + cure it. The code is wrong, clearly. In a situation where you set + the CDR's amaflags, and then ForkCDR, and then set the new CDR's + amaflags to some other value, you will see that all CDRs have had + their amaflags changed. This is not good. So I fixed it. ........ + +2007-06-21 21:40 +0000 [r70899] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 + lines Don't explode if the gain option is specified without a + value. (issue #9274 reported by mfarver) ........ + +2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant + + * channels/chan_iax2.c: Put the thread reading from the socket back + in the idle list if it deferred the processing of a full frame to + another thread + + * channels/chan_iax2.c: If a full frame is received while one of + the iax2 threads is in the middle of handling a full frame for + the same call, queue it up for processing by that same thread + later instead of dropping it. (issue #9937, patch by me) + +2007-06-21 20:19 +0000 [r70841] Steve Murphy + + * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 + line it was pointed out that the cdr_custom config load could get + a lock, and under certain circumstances, would never release it. + I also noted that the situation where more than one mapping spec + was warned about, but did not ignore further mappings as it had + promised. I think I have fixed both situations. ........ + +2007-06-21 19:49 +0000 [r70808] Mark Michelson + + * apps/app_voicemail.c: When volgain is used don't leave a + temporary file behind. (Closes Issue 8514, Reported and patched + by ulogic, code reviewed by Jason Parker) + +2007-06-21 15:22 +0000 [r70727] Joshua Colp + + * main/rtp.c: Do not Packet2Packet bridge if packetization settings + do not allow it. (issue #9117 reported by phsultan) + +2007-06-21 15:21 +0000 [r70726] Russell Bryant + + * apps/app_meetme.c: Remove a couple of duplicate unlocks + +2007-06-21 13:58 +0000 [r70677] Joshua Colp + + * apps/app_voicemail.c: Fix building with ODBC storage enabled. + (issue #10025 reported by denisgalvao) + +2007-06-21 13:00 +0000 [r70656] Steve Murphy + + * main/cdr.c: Via complaints aired in asterisk-users, I submit + these changes, which allow cdr updates to see macro + context/exten, whether hung up or not + +2007-06-20 23:32 +0000 [r70554-70612] Jason Parker + + * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql. + Issue 10020, patch by my, with credit to prashant_jois for + pointing out the problem. + + * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race + condition fix + + * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur + when reloading the module. Issue 10022, patch by me, with credit + to prashant_jois for finding the bug. + +2007-06-20 22:22 +0000 [r70552] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 + lines Don't overwrite the configured username setting upon a + REGISTER. (issue #8565 reported by jsmith) ........ + +2007-06-20 20:53 +0000 [r70494] Jason Parker + + * channels/chan_skinny.c: Make sure we clear the previously dialed + number if it did not exist. Issue 9958. + +2007-06-20 19:29 +0000 [r70445] Tilghman Lesher + + * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) + | 2 lines Issue 9997 - Timelimit times out the wrong channel + ........ + +2007-06-20 18:46 +0000 [r70397] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | + 5 lines Fix a problem where an established call would not be + properly disconnected when a PRI disconnect is received depending + on which cause code was received. (issue #9588, original patch by + softins, updated patch from jtexter3, and some additional + feedback from mhardeman) ........ + +2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp + + * main/rtp.c, main/frame.c: Put the speex packetization values back + in but disable it when setting up the smoother. + + * main/frame.c: Don't do packetization/smoother stuff with speex, + it doesn't work. + +2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant + + * contrib/scripts/ast_grab_core: don't delete the backtrace in + ast_grab_core + + * channels/chan_gtalk.c: Only attempt to queue a hangup on the + owner channel if it actually exists. (issue #9795, patch from + zandbelt) + +2007-06-19 18:23 +0000 [r70062] Steve Murphy + + * main/channel.c, /: Merged revisions 70053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 + line This fixes 9246, where channel variables are not available + in the 'h' exten, on a 'ZOMBIE' channel. The fix is to + consolidate the channel variables during a masquerade, and then + copy the merged variables back onto the clone, so the zombie has + the same vars that the 'original' has. ........ + +2007-06-19 17:07 +0000 [r70003] Joshua Colp + + * main/rtp.c, /: Merged revisions 69992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 + lines Handle the CC field in the RTP header. (issue #9384 + reported by DoodleHu) ........ + +2007-06-19 16:24 +0000 [r69987] Joshua Colp + + * main/channel.c, /: Merged revisions 69986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 + lines Update BRIDGEPEER variable if set to the new channel name + when a masquerade happens. (issue #9699 reported by dimas) + ........ + +2007-06-19 15:22 +0000 [r69944] Russell Bryant + + * channels/chan_sip.c: Fix a crash that could occur when handing + device state changes. When the state of a device changes, the + device state thread tells the extension state handling code that + it changed. Then, the extension state code calls the callback in + chan_sip so that it can update subscriptions to that extension. A + pointer to a sip_pvt structure is passed to this function as the + call which needs a NOTIFY sent. However, there was no locking + done to ensure that the pvt struct didn't disappear during this + process. (issue #9946, reported by tdonahue, patch by me, patch + updated to trunk to use the sip_pvt lock wrappers by eliel) + +2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 + lines Perform an extra hangup check just in case. (issue #9589 + reported by bcnit) ........ + + * /, res/res_features.c: Merged revisions 69846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 + lines Add parked call extension AFTER the parking slot has been + announced, otherwise two threads will try to handle the same + channel and it will go kaboom. (issue #9191 reported by japple) + ........ + + * main/callerid.c: Fix for building on PowerPC under Linux. + +2007-06-18 19:48 +0000 [r69796] Tilghman Lesher + + * channels/chan_sip.c: Issue 10005 - Segfault with missing + arguments, plus fix a missing define for SIP INFO channels + +2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp + + * channels/chan_sip.c: Don't count RTP timeout when involved in a + T38 fax session. (issue #9222 reported by ivoc) + + * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 + lines Set the peer name on the dialog to the one configured in + sip.conf and NOT the username to be used for authentication + attempts. (issue #9967 reported by achauvin) ........ + +2007-06-18 17:46 +0000 [r69744] Tilghman Lesher + + * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) + | 2 lines Issue 9998 - Remove SIG prefix, since it's not + supported by ksh ........ + +2007-06-18 16:51 +0000 [r69708] Joshua Colp + + * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called + for the first time so that it does not needlessly spit out + changed messages when the host really didn't change. + +2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant + + * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c, + build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, + include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c: + To prevent 92138749238754 more reports of "I have unixodbc + installed, but still can't build *_odbc.so!", check for ltdl + directly, instead of just listing it as another library to + include in the unixodbc check in the configure script. This also + makes ltdl show up as a dependency in menuselect so people know + what to go install. (related to issue #9989, patch by me) + + * build_tools/prep_moduledeps: Change the use of "echo -e" to + "printf". On systems where /bin/sh is not bash, most of the lines + in menuselect-tree were getting a "-e" at the beginning of every + line. I'm surprised nobody noticed this, but I think the XML + parser was being very nice and ignoring them. + +2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp + + * channels/chan_sip.c: Don't defer the BYE till later on a transfer + when the transfer itself goes kaboom and has no hope of working. + + * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock + something we never locked, and better handle a specific scenario + with doing an attended transfer between two non-bridged calls. + +2007-06-18 15:46 +0000 [r69660] Russell Bryant + + * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) + +2007-06-18 13:55 +0000 [r69625] Joshua Colp + + * channels/chan_sip.c: Fix issue where it would be possible for the + negotiated codecs to get set back to nothing. (issue #9992 + reported by yehavi) + +2007-06-15 Russell Bryant + + * Asterisk 1.4.5 released. + +2007-06-15 20:18 +0000 [r69579] Russell Bryant + + * res/res_features.c: Fix a silly deadlock in res_features that I + found while debugging on one of blitzrage's test machines. It was + one of the situations where he was seeing hung channels, and may + be the cause of some of the reports from other people. (related + to issue #9235) + +2007-06-15 19:23 +0000 [r69558] Joshua Colp + + * apps/app_speech_utils.c: Add support for setting the maximum + length of acceptable DTMF in SpeechBackground. + +2007-06-15 15:27 +0000 [r69518] Russell Bryant + + * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated + "SUCCESS" for both an answer of the incoming call on the trunk, + or if the trunk reached its ring timeout. This patch changes the + variable to say "RINGTIMEOUT" in that case. (issue #9973, + reported by n00dle, patch by me) + +2007-06-14 23:22 +0000 [r69434-69470] Jason Parker + + * main/config.c, /: Merged revisions 69469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 + lines Fix an issue where the line number in an unterminated + comment block error message would show the wrong line number. + "Reported" to me on #asterisk (somebody posted an error message, + and I happened to catch it) ........ + + * sounds/Makefile: Update to latest versions of sound files. + +2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming + + * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, + cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c, + main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c, + apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c, + main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, + channels/chan_iax2.c: use ast_localtime() in every place + localtime_r() was being used + +2007-06-14 21:08 +0000 [r69358] Russell Bryant + + * main/say.c: Fix some problems with saying dates and times for the + "tw" langauge (issue #9964, ljmid) + +2007-06-14 15:21 +0000 [r69259] Jason Parker + + * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun + 2007) | 4 lines Change a quite broken while loop to a for loop, + so "continue;" works as expected instead of eating 99% CPU... + Issue 9966, patch by me. ........ + +2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp + + * channels/chan_iax2.c: Whoops... + + * channels/chan_iax2.c: Let's make chan_iax2 media only native + transfers actually work. (issue #9376 reported by simone + cittadini) + + * channels/iax2-parser.c: Add TXMEDIA to list so that it is + properly displayed during iax2 packet output. + +2007-06-13 19:57 +0000 [r69183] Russell Bryant + + * channels/chan_sip.c: Move the logic for destroying a call when no + response is received to a BYE outside of the block that checks + for FLAG_FATAL to be set. This flag is only set when the packet + is transmitted with the reliability set to XMIT_CRITICAL when the + original packet is transmitted. A BYE is always sent with it set + to XMIT_RELIABLE, meaning this code could never be encountered. + This resulted in seeing some SIP channels that would never go + away with the last packet sent being a BYE. (part of issue #9235, + patch from jcmoore) + +2007-06-13 19:41 +0000 [r69181] Mark Michelson + + * apps/app_voicemail.c: Contains a patch for fixing an encoding + problem when using Outlook to view voicemail emails and + attachments. This fix has also been tested on Thunderbird, + Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, + patched by mutterc) + +2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp + + * apps/app_meetme.c: Really ignore NULL frames and check whether + the channel hungup or not. (issue #9912 reported by junky) + + * /, main/app.c: Merged revisions 69127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 + lines Return group counting to previous behavior where you could + only have one group per category. (issue #9711 reported by + irroot) ........ + +2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant + + * channels/chan_sip.c: Clarify a bit of logic. This doesn't change + behavior in any way, but it is helpful when following the logic + to debug problems like 9235. + + * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct + was accessed without the lock held. This issue was reported to me + via email by Dmitry Mishchenko. Thanks! + + * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois + in #asterisk-bugs. PQclear() was not called on the result + structure after doing a PQexec(). Also, fix up some formatting in + passing. + +2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp + + * channels/chan_iax2.c: Change the full frame dropping log message + to debug to avoid future bug reports. + + * channels/chan_iax2.c: Schedule the sending of a PING packet a + second later than previously so that it does not collide with the + LAGRQ. + +2007-06-12 19:13 +0000 [r69010] Russell Bryant + + * main/channel.c: In ast_channel_make_compatible(), just return if + the channels' read and write formats already match up. There are + code paths that call this function on a pair of channels multiple + times. This made calls fail that were using g729 in some cases. + The reason is that codec_g729a will unregister itself from the + list of available translators will all licenses are in use. So, + the first time the function got called, the right translation + path was allocated. However, the second time it got called, the + code would not find a translation path to/from g729 and make the + call fail, even if the channel actually already had a g729 + translation path allocated. (SPD-32) + +2007-06-12 14:23 +0000 [r68922] Joshua Colp + + * main/rtp.c, /: Merged revisions 68921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 + lines Bring RTP back to Asterisk at the end of a native bridge no + matter what. ........ + +2007-06-11 21:20 +0000 [r68814] Jason Parker + + * include/asterisk/time.h: Solaris 10 sometimes (?) needs this + include in order to have NULL defined. + +2007-06-11 20:45 +0000 [r68781] Tilghman Lesher + + * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly + used + +2007-06-11 16:57 +0000 [r68733] Christian Richter + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 68732 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | + 1 line added check for NULL Pointer when calling misdn_new. + Asterisk does not allow us to create channels anymore when stop + gracefully is used :). also modified the restart_indicator to 0 + ........ + +2007-06-11 14:33 +0000 [r68683] Joshua Colp + + * main/channel.c, /: Merged revisions 68682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 + lines Improve deadlock handling of the channel list. (issue #8376 + reported by one47) ........ + +2007-06-11 10:29 +0000 [r68644] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | + 1 line fixed problem that the dummybc chanels had no lock, + checking for the lock now. Also fixed the channel restart stuff, + we can now specify and restart particular channels too. ........ + +2007-06-11 04:21 +0000 [r68595] Tilghman Lesher + + * pbx/pbx_config.c: "dialplan save" produced garbage in the config + file + +2007-06-08 22:23 +0000 [r68527] Russell Bryant + + * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | + 4 lines Don't automatically hang up after running Dictate so that + callers can exit cleanly using '#' (closes issue #9577, patch + from Thomas Andrews) ........ + +2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming + + * channels/chan_iax2.c: actually remember the type/subclass of full + frames that are in process + +2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp + + * /, main/say.c: Merged revisions 68397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 + lines Don't call ast_waitstream_full when the control file + descriptor and audio file descriptor are not set, simply call + ast_waitstream! (issue #8530 reported by rickead2000) ........ + + * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 + lines Do a DNS lookup immediately upon calling the dnsmgr + function, don't wait until a refresh happens. (issue #9097 + reported by plack) ........ + +2007-06-07 23:14 +0000 [r68354] Russell Bryant + + * /, main/say.c: Merged revisions 68351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | + 3 lines Fix a problem where saying a character wouldn't properly + break out when the caller pressed '#' (issue #8113, reported by + patbaker82, patch from jamesgolovich (hey, long time no see!) and + patbaker82) ........ + +2007-06-07 23:00 +0000 [r68326] Jason Parker + + * apps/app_voicemail.c: Fix incorrect French syntax of "old + messages". Request for feedback was sent to asterisk-dev mailing + list, with little response. Issue 9118, patch by junky. + +2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming + + * channels/chan_iax2.c: some improvements to the IAX2 full frame + dropping logic recently added: - use inaddrcmp(), since we have + it - output the type of frame and subclass being dropped, and the + type/subclass that is already being processed (which caused the + drop) + +2007-06-07 21:16 +0000 [r68280] Russell Bryant + + * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent + queue members when using realtime configuration for queues. Also, + remove an unneeded leading slash for the astdb family. (issue + #9911, patch by atis) + +2007-06-07 20:25 +0000 [r68211-68249] Jason Parker + + * channels/chan_skinny.c: Fix an issue with newer phones which + require packets be padded out to the correct length. Issue 9887, + patch by DEA. + + * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 + lines Don't try to save voicemail greetings unless the user + presses '1' to accept/save. Issue 9904, patch by me. ........ + +2007-06-07 19:47 +0000 [r68198] Mark Michelson + + * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a + check to make sure that greetings get stored properly. (Issue + 8016, reported by edhorton, patched by alamantia with + modification by me. Thanks to Jason Parker for the advice on + this). + +2007-06-07 19:46 +0000 [r68196] Olle Johansson + + * channels/chan_features.c: Disable chan_features by default in + menuselect + +2007-06-07 19:30 +0000 [r68192] Russell Bryant + + * main/strcompat.c: Include stdarg.h for build issues on Solaris + (issue #9381) + +2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp + + * main/channel.c: Fix logic when doing a name based channel search + for a structure when you want to start from a specific point in + the channel list. (issue #9324 reported by slavon) + + * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 + lines Allow the 'g' option to work if used with the 'S' option. + (issue #9888 reported by gasparz) ........ + +2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson + + * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't + forget. + + * res/res_jabber.c: Ok, we found out that this is not about if you + have any *active* clients using TLS, but if you have initialized + TLS at all during the lifetime of the module. So if you reload to + disable TLS, it won't help. + + * res/res_jabber.c: If you have a jabber client that uses TLS, + refuse unload. Bad fix, but will prevent crashes while we are + trying to find a workaround. Iksemel development seems to have + stalled and we might have to stop using the TCP/TLS connections + in that library and use our own, which would scale better from a + poll/select perspective I guess. It would also make it easier to + migrate to OpenSSL and stop Asterisk from depending on both + OpenSSL and GnuTLS. + + * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make + sure we can unload res_jabber. Patch by phsultan - thanks! Due to + a bug in the iksemel library, this will not work if you are using + GTLS in the connection. That's being investigated. If you figure + out a way to handle that without us having to patch iksemel, let + us know in the bug report. Thanks. + +2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 + lines Only notify the devicestate system of a peer state change + when the peer is built from the config file. (issue #9900 + reported by arkadia) ........ + + * main/file.c: Properly handle cases where a stream can't be + written to. (issue #9757 reported by junky) + +2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant + + * res/res_snmp.c: Disable reload functionality in res_snmp. It is + not possible to initialize the snmp library more than once + without completely unloading the module and loading it again. + (issue #9571, reported by hristo, additional helpful debug + information from festr, patch from me) + + * channels/chan_sip.c: Fix a crash when doing call pickups with SIP + phones. The code unlocked the channel when it should not have. + (issue #9652, reported by corruptor, fixed by me) + +2007-06-06 19:26 +0000 [r67804] Mark Michelson + + * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault + under a specific set of circumstances: 1. VoiceMailMain was + configured in the dialplan with an extension as its argument 2. A + message was left for this mailbox 3. Tried to call VoiceMailMain + but hung up before entering password. This was fixed by checking + that a pointer was non-null prior to trying to dereference it. + (Issue 9810, reported by xmarksthespot, patched by Corydon76 with + modifications by me). + +2007-06-06 16:55 +0000 [r67716] Russell Bryant + + * main/channel.c, /, include/asterisk/linkedlists.h: Merged + revisions 67715 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | + 5 lines We have some bug reports showing crashes due to a double + free of a channel. Add a sanity check to ast_channel_free() to + make sure we don't go on trying to free a channel that wasn't + found in the channel list. (issue #8850, and others...) ........ + +2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp + + * main/rtp.c, /: Merged revisions 67649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 + lines Reinvite the RTP back to the Asterisk machine when the + timeout happens. (issue #9888 reported by gasparz) ........ + + * main/translate.c: Fix plc_samples warning when registering a + translator. (issue #9897 reported by xylome) + + * apps/app_directed_pickup.c: Include macroexten while searching + for a channel to pick up in case they are in a macro. (issue + #9491 reported by jamesb63) + + * res/res_agi.c: Make the new "agi debug off" CLI command work. + (issue #9890 reported by eliel) + + * /, main/devicestate.c: Merged revisions 67593 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 + lines Revert channel name splitting fix for Zap. The moral of the + story is don't use - in your user/peer names. (issue #9668 + reported by stevedavies) ........ + +2007-06-05 23:01 +0000 [r67558] Russell Bryant + + * apps/app_meetme.c: Fix some crashes related to the use of the + "meetme" CLI command. The code for this command was not locking + the conference list at all. (issue #9351, reported by and patch + submitted by Junk-Y, committed patch is different and by me) + +2007-06-05 21:30 +0000 [r67526] Steve Murphy + + * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug + 9883, wherein macros were not allowing the includes construct. + fixed and tested, looks OK. Now includes can serve as an adjunct + to catch. + +2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant + + * include/asterisk/linkedlists.h: This bug has been hanging over my + head ever since I wrote this SLA code. Every time I tried to go + debug it by adding some debug output, the behavior would change. + It turns out I wasn't crazy. I had the following piece of code: + if (remove) AST_LIST_REMOVE_CURRENT(...); Well, + AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my + conditional statement didn't do much good at all. It always ran + at least all of the macro minus the first statement, so I was + seeing list entries magically disappear when they weren't + supposed to. After many hours of debugging, I have come to this + extremely irritating fix. :) (issues #9581, #9497) + + * channels/chan_zap.c: Suppress a bunch of debug output unless + option_debug is on + +2007-06-05 18:32 +0000 [r67424] Mark Michelson + + * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails + saved to IMAP storage using extensions other than gsm were unable + to be played over the phone. (Issue 9786, reporter: + xmarksthespot, Patched by xmarksthe spot with revisions by me, + reviewed by Russell Bryant). + +2007-06-05 18:18 +0000 [r67421] Jason Parker + + * channels/chan_skinny.c: Correctly update date/time on devices + throughout the life of the device, instead of just at + registration. Issue 9152, yet another patch by DEA. + +2007-06-05 18:17 +0000 [r67420] Steve Murphy + + * pbx/pbx_ael.c: Added code to automatically add a default case to + switches that don't have one. In some cases, rather than fall + thru, it results in a goto with -1 result, which terminates the + extension; a sort of dialplan seqfault, sort of. This was + required to fix bug reported in 9881 + +2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant + + * main/channel.c: Handle a failure in malloc() in + ast_safe_string_alloc() + + * main/channel.c: Fix a problem that showed itself by causing Zap + channel names to be completely bogus on my machine. + ast_safe_string_alloc() was broken. It called vsnprintf() on a + va_args list twice without re-initializing it. After the first + usage, va_end() and va_start() must be called again. + +2007-06-05 16:14 +0000 [r67329-67334] Christian Richter + + * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | + 1 line briding is a bool, fixed copy and paste issue. ........ + + * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 + Jun 2007) | 1 line simplified the EVENT_SETUP handling in the + cb_events function a lot. Commented the different possibilities a + bit and made functions of shared code. When the dialed extension + does not exist in the extensions.conf we'll jump into the 'i' + extension if this does exist, else we disconnect the call with + the cause:1 = No Route to Destination. ........ + +2007-06-05 15:51 +0000 [r67308] Russell Bryant + + * main/asterisk.c, main/loader.c, include/asterisk/module.h: When + shutting down "gracefully", go through and run the unload() + callbacks for all of the modules. "stop now" is considered a + non-graceful shutdown and will not go through this process. + (issue #9804, reported by chrisost, patch by me) + +2007-06-05 15:22 +0000 [r67304] Joshua Colp + + * channels/chan_iax2.c: Only muck with the thread structure if an + idle one was found/created. + +2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming + + * channels/chan_iax2.c: ensure that a burst of full frames + (AST_FRAME_DTMF being the prime example) will not be processed + out of order... this is a brute force fix, but seems to be the + safest fix for now (thanks to the Digium PQ department for + finding this bug) + +2007-06-05 10:25 +0000 [r67210] Christian Richter + + * channels/misdn_config.c, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h: Merged revisions 67209 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | + 1 line added possibility to deactivate bridging per port ........ + +2007-06-04 23:43 +0000 [r67162] Tilghman Lesher + + * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) + | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. + ........ + +2007-06-04 23:31 +0000 [r67158] Russell Bryant + + * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt + structure can disappear and the code did not account for it and + crashes. (issues #9642, #9569, #9666, probably others ... based + on the work by stevedavies and mihai, with additional changes + from me) + +2007-06-04 23:26 +0000 [r67121-67156] Jason Parker + + * channels/chan_skinny.c: Fix for skinny keepalives. If there is no + traffic from the phone for (keep_alive * 1100) ms (arbitrarily + adding 10% for network issues, etc), unregister the device. Issue + 8394, patch by DEA. + + * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar + to the recent fix for chan_skinny) Issue 9855, patch by DEA. + +2007-06-04 22:28 +0000 [r67119] Russell Bryant + + * channels/chan_iax2.c: Add comments for two functions that get + called with the appropriate call locked, but perform operations + that could result in the pvt structure getting destroyed before + returning again, causing numerous seg faults all over the module. + (inspired by issues #9642, #9569, and #9666, and the work done by + stevedavies and mihai) + +2007-06-04 21:59 +0000 [r67073] Steve Murphy + + * main/cdr.c: This typo has been here since 1.4 forked. It has been + the source of heartburn to many a dialplan/CDR programmer. + +2007-06-04 21:47 +0000 [r67071] Russell Bryant + + * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) + +2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp + + * channels/chan_sip.c: Better handle SIP devices that say they have + SDP content... but really don't. (issue #9398 reported by + mthomasslo) + + * apps/app_dial.c: Initialize cidname variable to nothing since it + may be used without having been touched. (issue #9661 reported by + dimas) + + * res/res_features.c: Returning a value that indicates the parking + of a call was a success when it really wasn't (because the + parking slot selected was in use) is the wrong thing to do. + (issue #9723 reported by mdu113) + +2007-06-04 17:11 +0000 [r67061] Tilghman Lesher + + * contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, /, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.mandrake.zaptel, + contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) + | 2 lines Add revision Id tags (by request of tzafrir) ........ + +2007-06-04 16:02 +0000 [r67026] Russell Bryant + + * configure, configure.ac: Change the configure script to build a + test program against libcurl to make sure the results from + curl-config can be used to compile successfully. This is intended + to help prevent a situation where you are cross compiling, and + the configure script finds the curl library installed on the + host. (issue #9865, reported and patched by zandbelt) + +2007-06-04 15:50 +0000 [r67021] Tilghman Lesher + + * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash + +2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant + + * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When + handling an implicit ACK to a frame that was marked as the final + transmission for a call, don't call iax2_destroy() for that call + while the global frame queue is still locked. There is a very + nice explanation of the deadlock in the report. (issue #9663, + thorough report and patch from stevedavies, additional positive + test reports from mihai and joff_oconnell) + + * include/asterisk/stringfields.h: Fix some compiler warnings in + C++ modules. (issue #9866, reported by osk, patch by Corydon76) + +2007-06-01 21:45 +0000 [r66919] Tilghman Lesher + + * funcs/func_odbc.c: On some drivers, deallocating the statement + handle isn't enough. We also have to clear the cursor (nice, + Oracle) + +2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson + + * apps/app_voicemail.c: Removing extraneous debugging lines from + revision 66897. Sorry :) + + * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP + storage. Attachments with format specified as gsm were duplicated + (i.e. two attachments) were left. Thank you very much to + xmarksthespot for submitting the patch that fixed this. (Issues + 9787 and 8873, Reported by xmarksthespot and jerjer, patched by + xmarksthespot) + +2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant + + * channels/chan_skinny.c: Changes to the way DTMF is handled in the + core broke dialing in chan_skinny. This patch makes chan_skinny + usable again. I did not end up testing this, but there are + multiple positive test reports listed in the bug report. (issue + #9596, reported by pj, testing by pj and mvanbaak, and the fix + was written by DEA) + + * apps/app_page.c: List app_meetme as a module that app_page + depends on. + +2007-05-31 23:03 +0000 [r66821] Tilghman Lesher + + * doc/asterisk.8: Issue 9850 - update preferred command line syntax + +2007-05-31 18:41 +0000 [r66775] Russell Bryant + + * res/res_speech.c, include/asterisk/app.h, + include/asterisk/speech.h: Change a couple of header files to not + use "new", which is a reserved keyword in C++. (issue #9830, + reported by osk) + +2007-05-31 17:15 +0000 [r66770] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) + | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. + Issue 8329 will remain unfixed for pbx_realtime, but only because + we lack core API to do it. ........ + +2007-05-31 16:14 +0000 [r66768] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 + lines It is now possible for this path of execution to have the + frame pointer be NULL, therefore we need to check for it before + trying to access it. (issue #9836 reported by barthpbx) ........ + +2007-05-30 23:26 +0000 [r66671] Mark Michelson + + * apps/app_voicemail.c: Fixed seg-faults when recording greetings + in voicemail with IMAP enabled. (Issue No. 9735, reported by + xmarksthespot, patched by me) + +2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp + + * channels/chan_sip.c: Silly me for having out of date source! Oh + well... I'm still leaving my comment. + + * channels/chan_sip.c: When calling some peer/host that may not + exist/reply back... don't keep the dialog in memory for all of + eternity. + + * channels/chan_zap.c, channels/chan_features.c: Change how channel + names are generated a bit. (issue #9825 reported by eldadran) + +2007-05-29 21:56 +0000 [r66538] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) + | 2 lines If the value of a variable passed to FIELDQTY is blank, + then FIELDQTY should return 0, not 1. ........ + +2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson + + * channels/chan_sip.c: Properly handle 408 request timeout - + according to the RFC, the dialog dies if a request in a dialog + gets this response. + + * channels/chan_sip.c: Don't issue hangup on hangup on hangup on + hangup (for jcmoore) + +2007-05-29 16:44 +0000 [r66437] Joshua Colp + + * main/rtp.c: Handle cases where a frame may have no data. (issue + #9519 reported by dmb) + +2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson + + * channels/chan_sip.c: Don't reset hangupcause if we already have + one + + * channels/chan_sip.c: Tracking down hanging channels, killing them + one by one. Issue #9235 and related + +2007-05-29 15:43 +0000 [r66398] Joshua Colp + + * doc/datastores.txt: Update datastores documentation. (issue #9801 + reported by mnicholson) + +2007-05-29 09:41 +0000 [r66363] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 + lines Issue #9802 - Change inuse counter on CANCEL ........ + +2007-05-28 23:16 +0000 [r66312] Joshua Colp + + * channels/chan_zap.c: Make the usedistinctiveringdetection option + work again. (issue #9823 reported by premeau) + +2007-05-27 04:12 +0000 [r66244] Jason Parker + + * channels/chan_zap.c: I don't know what this was trying to do, but + it's clearly incorrect. Issues 9808 and 9809. + +2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming + + * configure, configure.ac: have to check for OSP toolkit _after_ + checking for OpenSSL + +2007-05-25 14:41 +0000 [r66159] Tilghman Lesher + + * /, main/say.c: Merged revisions 66127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) + | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch + ........ + +2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming + + * configure, configure.ac, channels/chan_gtalk.c, makeopts.in, + res/res_jabber.c: handle the GNUTLS library properly in the + configure script and build system don't build in OSP support + unless we have found and are allowed to use SSL support + +2007-05-24 22:23 +0000 [r66076] Russell Bryant + + * main/channel.c: if the string field init fails, clean up the + stuff that was allocated already + +2007-05-24 22:16 +0000 [r66074] Joshua Colp + + * main/slinfactory.c: Fix slinfactory logic when dealing with + frames coming in that may already be in the signed linear format. + +2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant + + * main/channel.c: Check the result of ast_string_field_init() in + ast_channel_alloc() + + * main/rtp.c: Make 1.4 build on my machine, too.. + +2007-05-24 20:54 +0000 [r66029-66030] Jason Parker + + * configure: Rebuild configure script for previous ar fix. + + * configure.ac: Following moving strip to AC_PATH_TOOL, we need to + do something similar for ar. + +2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Checking for the strip application needs to be done with + AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross + compilation environments. + + * Makefile: Clear CFLAGS before running make for menuselect. (issue + #9784, reported by ovi, patch by me) + +2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming + + * channels/chan_gtalk.c: oops, use #ifdef instead of #if + + * channels/chan_gtalk.c: don't reference GnuTLS headers and + functions unless the configure script found it + + * main/rtp.c: don't use uninitialized variables + +2007-05-24 15:27 +0000 [r65902] Joshua Colp + + * main/manager.c: Add the ability to blacklist certain commands + from being executed using the Command AMI action. (issue #9240 + reported by junky) + +2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson + + * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk + core dump since the GnuTLS interface did not support + multithreading correctly. + + * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN. + Patch by phsultan. Thanks! + +2007-05-24 15:16 +0000 [r65877-65883] Jason Parker + + * .cleancount: Update cleancount for that last commit - just for + good measure. + + * include/asterisk/translate.h, codecs/codec_speex.c, + main/translate.c, codecs/codec_ilbc.c: Fix handling of + zero-length frames when a codec is capable of native PLC. Issue + 9183, patch by Mihai. + +2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard + + * funcs/func_math.c: merged qwell's func_math patch for issue 9507 + +2007-05-24 15:08 +0000 [r65863] Joshua Colp + + * main/rtp.c: I like it when the RTP stack compiles myself... + +2007-05-24 15:05 +0000 [r65857] Olle Johansson + + * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues + when calling from gtalk to SIP over nat. + +2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant + + * apps/app_festival.c: Ensure that frames are fully initialized. + This will probably fix getting weird timestamp log messages in + logs when using the Festival app. (issue #9781, patch by me) + + * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous + code would result in oscillating and incorrect data. + Additionally, the RTT would sometimes report negative values due + to incorrect calculations. (issue #9601, patch from davetroy) + +2007-05-24 14:48 +0000 [r65841] Olle Johansson + + * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for + jingle + +2007-05-24 14:42 +0000 [r65839] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 + lines Allow RFC2833 to be negotiated when an INVITE comes in + without SDP and is not matched to a user or peer. (issue #9546 + reported by mcrawford) ........ + +2007-05-24 14:38 +0000 [r65836] Olle Johansson + + * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan - + Fix "login" as component to jabber server. ...and, by accident, + fix a bug in chan_sip for stopping a loop on retransmits of BYE + requests. + +2007-05-24 09:37 +0000 [r65768] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 + Mai 2007) | 1 line we should only activate the generator in + chan_misdn, when asterisk hask not yet taken the call + (WAITING4DIGS state). Alerting audio will be generated fomr + asterisk for example. ........ + +2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming + + * channels/chan_iax2.c: start the delayed PBX when receive voice or + video full frames as well, and comment this delayed-PBX activity + + * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) + | 2 lines ensure that variables are set on a newly created + channel before we start a PBX on it ........ + + * channels/chan_iax2.c: clear the 'delay PBX' flag when we are + ready to start the PBX + + * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2 + channel until we have some sort of response to our ACCEPT (ACK or + anything else) + + * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) + | 2 lines if we are going to set variables on a newly created + channel, it should be done *before* we start the PBX on it + ........ + +2007-05-23 13:07 +0000 [r65589] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | + 3 lines Revert revision 62417 as someone reported problems with + it to Mark. This was related to issue #9588. ........ + +2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming + + * build_tools/make_version: when building a version string for a + developer branch, include the base branch in the version string + +2007-05-22 18:40 +0000 [r65501] Russell Bryant + + * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a + dependency for app_voicemail and chan_zap (Thanks to mnicholson + for pointing it out) + +2007-05-22 15:04 +0000 [r65452] Joshua Colp + + * apps/app_meetme.c: Remove a double const. + +2007-05-22 14:02 +0000 [r65408] BJ Weschke + + * apps/app_followme.c: Fix a problem with flag recognition. + +2007-05-22 13:09 +0000 [r65394] Russell Bryant + + * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | + 4 lines Fix a memory leak that I just noticed in the device state + handling in app_queue. On most device state changes, it would + leak roughly 8 to 64 bytes (the length of the name of the + device). ........ + +2007-05-22 08:12 +0000 [r65342] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 + Mai 2007) | 1 line we stop the tones only when we're in the + pre-call phase, otherwise e.g. when in CONNECTED state we should + not stop tones when we receive an Information Message ........ + +2007-05-20 17:59 +0000 [r65250] Joshua Colp + + * res/res_agi.c: res_agi needs to export two symbols + (ast_agi_register and ast_agi_unregister) for usage by others. + (issue #9755 reported by mnicholson) + +2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy + + * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c + to main/cdr.c, and neither did I. This is the remainder of the + 9717 patch, the fix for the run-away FAIL status for a call + + * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions + 65172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 + line This update will fix the situation that occurs as described + by 9717, where when several targets are specified for a dial, if + any one them reports FAIL, the whole call gets FAIL, even though + others were ringing OK. I rearranged the priorities, so that a + new disposition, NULL, is at the lowest level, and the + disposition get init'd to NULL. Then, next up is FAIL, and next + up is BUSY, then NOANSWER, then ANSWERED. All the related set + routines will only do so if the disposition value to be set to is + greater than what's already there. This gives the intended + effect. So, if all the targets are busy, you'd get BUSY for the + call disposition. If all get BUSY, but one, and that one rings is + not answered, you get NOANSWER. If by some freak of nature, the + NULL value doesn't get overridden, then the disp2str routine will + report NOANSWER as before. ........ + +2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 + lines Not getting an ACK to a 200 OK in the initial invite is + critical to the call. ........ + + * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 + lines Issue 9235 - part of the problem, maybe not all. Please + retry with this patch (and no other patch) if you have problems + with hanging SIP channels. Thank you. A special Thank You to + WeBRainstorm that gave me access to his system. ........ + + * channels/chan_sip.c: - Adding support for putting calls OFF hold + with a re-invite with blank SDP. This was a bug found while doing + tests at SIPit in Antwerp. - In order to not duplicate code, I + restructured some of the code for putting calls on/off hold. + Thanks DEA for reminding me. This fix has been asleep in the + videocaps branch until now. + +2007-05-18 12:40 +0000 [r65039] Christian Richter + + * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged + revisions 65007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | + 1 line fixed a warning regarding Keypad encoding. encode the IE + sending_complete at the right position. ........ + +2007-05-18 10:37 +0000 [r64974] Olle Johansson + + * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of + call + +2007-05-18 08:58 +0000 [r64904] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 + Mai 2007) | 1 line we *need* to send a PROCEEDING when + sending_complete is set, even if need_more_infos is requested. + ........ + +2007-05-18 02:48 +0000 [r64868] Russell Bryant + + * apps/app_queue.c: Fix a small bug I noticed while working on + something else. app_queue did not unregister its device state + monitoring callback in unload_module(). So, this would make + Asterisk crash on the first device state change after you unload + the module. + +2007-05-17 21:19 +0000 [r64820] Tilghman Lesher + + * /, include/asterisk/linkedlists.h: Merged revisions 64819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) + | 2 lines How is it that we never caught that this is returning + the opposite of our documentation, until now? ........ + +2007-05-17 16:53 +0000 [r64761] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 + lines If we have a negative current message, we shouldn't go back + even further... Issue 9727. ........ + +2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant + + * contrib/scripts/astxs (removed): Remove script that is no longer + functional since the build system was redone. (issue #9340, + reported by junky) + + * apps/app_dial.c: Increase the size of a buffer to support longer + dial strings for channels. (issue #9291, reported and fix + suggested by meni) + +2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp + + * channels/chan_sip.c: Even more direct RTP setup fixes! Don't + allow a codec that isn't supported to creep into the SDP of + either side. (issue #9446 reported by marcelbarbulescu) + + * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported + by xmarksthespot) + +2007-05-17 06:13 +0000 [r64686] Russell Bryant + + * README: Update the main README to reflect the new build process + for 1.4 and above. (issue #9725, patch by eliel) + +2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson + + * /: Blocking patch already in this code + + * channels/chan_sip.c: Fix auth on BYE. (Different patch than for + 1.2) + + * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE + + * channels/chan_sip.c: Final part of issue #9483 - fixing + transfer() of sip calls in the dial plan (twilson) + + * channels/chan_sip.c: Issue #9439 - properly handle username + parameters in SIP uri. + + * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 + lines Support SIP uri's starting with SIP: and sip: (reported by + Tony Mountfield on the mailing list. Thanks!) ........ + + * /, channels/chan_sip.c: Merged following patch with a lot of + changes for 1.4 ------ Merged revisions 64514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 + lines Issue #9726 - rlister - Better logging for ACL denials + While at it, also added better logging and handling of peers that + are not supposed to register. My patch, stole the issue report + from Russell. My apologies, Russell :-) ........ + +2007-05-16 08:44 +0000 [r64515] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 + Mai 2007) | 1 line in the case immediate=yes, we directly jump + into the dialplan, where people can use PlayTones to indicate a + Dialtone, so we don't need to to that by ourself. also we should + not do a dialtone_indicate for incoming calls on a TE port in + overlapdialmode. ........ + +2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant + + * res/res_features.c: Properly fix a problem that occurs when you + set PARKINGEXTEN to an exten where a call is already parked. + (issue #9723, patch by me) + + * res/res_features.c: When someone requests a specific parking + space using the PARKINGEXTEN variable, ensure that no other + caller is already there. (issue #9723, reported by mdu113, patch + by me) + +2007-05-14 19:26 +0000 [r64324] Olle Johansson + + * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit + more + +2007-05-14 19:13 +0000 [r64306] Russell Bryant + + * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by + just ignoring it. An unknown indication will trigger an error and + cause sounds to stop, which in this case, is ringing. + +2007-05-14 18:52 +0000 [r64280] Olle Johansson + + * channels/chan_sip.c: Handle network errors, like host or network + unreachable, in a better way. This means that calls to hosts or + qualify (OPTION) messages will fail quicker if the TCP/IP stack + tells us that there is an issue. Since this is an unconnected UDP + socket, we will not get error messages directly in most cases, + but maybe on the second and third try. This is already + implemented in trunk. + +2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp + + * codecs/codec_speex.c: Properly set datalen field when doing PLC + in codec_speex. (issue #9722 reported by mihai) + + * /, main/devicestate.c: Merged revisions 64275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 + lines Only perform stripping of - strings from the channel name + for Zap channels. Anywhere else we might remove a legitimate part + of a device name. (issue #9668 reported by stevedavies) ........ + + * main/channel.c: Fix scenario where if a phone that simply called + Echo() put itself on hold it could never get off hold. + +2007-05-14 13:58 +0000 [r64193] Steve Murphy + + * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570, + worrisome CDR warnings have been removed, that are either not + helpful, or not relevant. + +2007-05-14 10:39 +0000 [r64157] Olle Johansson + + * main/channel.c: Add hangupcause when we lack codecs for + transcoding + +2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp + + * channels/chan_sip.c: This concludes my final adventure with + bitmasks and the onhold flag. Would anyone care for some peanuts? + + * channels/chan_sip.c: Tweak hold flags some more. They can be of + three states when active: active, inactive, one direction. + + * channels/chan_sip.c: Ensure the onhold flag is set no matter what + when being put on hold. + +2007-05-11 20:16 +0000 [r63982] Jason Parker + + * main/manager.c: Hide manager password from "manager show user + foo". I realize that there are other ways to get this, but we + really don't need to just show it in plain text so easily. Issue + 9273, patch by junky + +2007-05-11 16:35 +0000 [r63905] Tilghman Lesher + + * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions + 63903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) + | 2 lines Issue 9121 - fixups for safe_asterisk script ........ + +2007-05-11 16:05 +0000 [r63886] Russell Bryant + + * main/manager.c: When MD5 authentication is not possible because + there is no challenge present, either because the Challenge + action was never issued, or some other reason, give a proper + error message and return an error instead of claiming that the + user wasn't found. (reported by jsmith on IRC) + +2007-05-11 15:43 +0000 [r63872] Joshua Colp + + * res/res_features.c: Make the PARKINGEXTEN feature of parking + actually work. (issue #9708 reported by mdu113) + +2007-05-10 23:15 +0000 [r63830] Jason Parker + + * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 + lines Fix an issue with trying to kill a thread before it gets + created. Issue 9709, patch by nic_bellamy. ........ + +2007-05-10 22:23 +0000 [r63804] Russell Bryant + + * main/manager.c: Strip terminal escape sequences from CLI command + output that is going to be sent out over the manager interface. + (issue #9659, reported by pari, fixed by me) + +2007-05-10 20:48 +0000 [r63750] Doug Bailey + + * main/callerid.c: Add test for negative offsets in cid data to + prevent infinite loops. + +2007-05-10 20:46 +0000 [r63749] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 + lines Do not allocate SIP pvt's for PEERs we can not reach. This + was seen as a lot of dialogs being created then immediately + destroyed at reload/restart of the SIP channel. ........ + +2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp + + * main/channel.c: Use the DTMF frame on the channel when returning + a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. + + * channels/chan_sip.c: Do not prematurely go on hold if sendonly + was not actually set. + +2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson + + * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 + lines Make sure we only create a DSP if it's requested on + SUB_REAL ........ + +2007-05-09 16:55 +0000 [r63612] Russell Bryant + + * main/channel.c: Modify ast_senddigit_begin() to use the same + assumptions used elsewhere in the code in that if a channel does + not have a send_digit_begin() callback, it only cares about DTMF + END events. (pointed out by Michael Neuhauser on the asterisk-dev + list) + +2007-05-09 16:54 +0000 [r63611] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 + lines Properly handle hints that point to multiple devices in + chan_sip. Why chan_sip is even doing this I have no idea but I + would rather not go into a rant. (issue #9536 reported by + rlister) ........ + +2007-05-09 16:43 +0000 [r63608] Russell Bryant + + * main/channel.c: Only call ast_senddigit_begin() in + ast_senddigit() if the channel has a send_digit_begin() callback. + Checking the END_DTMF_ONLY flag was the wrong thing to do, + because that flag indicates that a *bridged* channel only wants + DTMF END events coming from this channel. + +2007-05-09 14:50 +0000 [r63566] Tilghman Lesher + + * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) + | 2 lines Replicate fix from 51158 (app_voicemail) to + app_directory (Issue 9224) ........ + +2007-05-09 13:24 +0000 [r63535] Russell Bryant + + * Makefile: I have seen multiple people post questions trying to + figure out what the message "The configure script must be + executed before running 'make'" means. So, add another like that + says to specifically run ./configure. If this isn't obvious + enough, then they should be using something like AsteriskNOW and + not installing from source. + +2007-05-09 13:17 +0000 [r63534] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c: Merged revisions + 62945,63402,63519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | + 1 line when we're in state WAITING4DIGS, we use the asterisk + tone-generator which prods us, so we can't just return -1 in + misdn_write in this case. Added a MISDN_KEYPAD channel variable, + and fixed the sending of keypad. this enables us to modify the + call forward parameters in the switch. ........ r63402 | crichter + | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added + application misdn_check_l2l1 which tries to pull up the L1/L2 on + all ports that have the layers down in a group. It waits then for + a timeout. This helps for scenarios where multiple PMP BRIs are + grouped together, or where a provider has a faulty PTP + Implementation, that looses the L2 after a while. ........ r63519 + | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line + release_chan frees ch, so we should never touch ch after + release_chan, this may cause segfaults. ........ + +2007-05-09 13:04 +0000 [r63532] Olle Johansson + + * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status. + (Reported on asterisk-users) + +2007-05-08 22:38 +0000 [r63478] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) + | 2 lines Issue 9602 - segfault in app_macro ........ + +2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant + + * res/res_features.c: I mixed up the use of the find_feature() + function, so I renamed it find_dynamic_feature, and changed the + code to use the correct lock when using it. + + * res/res_features.c: Use a read/write lock when accessing the + built-in features. + + * contrib/scripts/realtime_pgsql.sql (added), + contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to + contrib/scripts to be with the rest of the sql examples. (issue + #9676, suretec) + +2007-05-08 06:22 +0000 [r63360] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) + | 2 lines Issue 9527 - upon entering a folder, no message is + selected (curmsg == -1), so deleting causes memory corruption + (beyond bounds) ........ + +2007-05-07 22:28 +0000 [r63329] Russell Bryant + + * configs/res_pgsql.conf.sample (added), + configs/extconfig.conf.sample, contrib/realtime_pgsql.sql + (added): Add a sample configuration file and example tables for + use with res_config_pgsql. (issue #9676, suretec) + +2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp + + * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged + revisions 63285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 + lines Properly handle what happens during a masquerade in + relation to group counting. (issue #9657 reported by ramonpeek) + ........ + + * channels/chan_sip.c: Minor backport of revision 59083 in trunk. + Don't queue an unhold frame up if the call was never on hold to + begin with. + +2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson + + * main/config.c: Don't remove configuration from memory just + because one section failed. + + * /: Guess svnmerge doesn't handle files that move around. Blocking + patch to ./config.c + +2007-05-06 12:28 +0000 [r63152] Olle Johansson + + * main/file.c: Stop the video stream when you stop playback of all + streams for a call + +2007-05-04 20:03 +0000 [r63099] Jason Parker + + * res/res_jabber.c: Fix a crash when checking version attribute in + an incoming XML caps element. Issue 9667, patch by phsultan. + +2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni + + * configs/manager.conf.sample: explanation for httptimeout in + manager.conf + +2007-05-03 16:44 +0000 [r62989] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 + lines When a peer is seeded or built tell the devicestate core to + update it's status. This is easier then having chan_sip load + before pbx_config. (issue #9658 reported by dlynes) ........ + +2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming + + * main/loader.c: improve loader a bit, by avoiding trying to + initialize embedded modules twice and avoiding trying to load + modules from disk when they have been loaded already during the + 'preload' pass (reported by blitzrage on IRC, patch by me) + +2007-05-03 15:23 +0000 [r62942] Russell Bryant + + * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell + Bryant, 2007, TM, Patent Pending). This set of changes came from + a debugging session I had with Dwayne Hubbard. When he called + into his home FXO, ran the Echo application, and pressed a digit, + the digit would be echoed back and would never end. This is + fixed, along with a couple other little improvements. * When + chan_zap is in the middle of playing a digit to a channel, it + feeds back null frames, not voice frames. So, I have modified + ast_read to check the timing on emulated DTMF when it receives + null frames, in addition to where it was doing this on voice + frames. * Make a tweak to setting the duration on emulated DTMF + digits. If there was no duration specified, it set it to be the + minimum, instead of the default. * Instead of timing the emulated + digits off of the number of samples in audio frames that pass + through, just use time values. Now there is no code in this + section that assumes 8kHz audio. + +2007-05-03 14:41 +0000 [r62913] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19 + (added), pbx/ael/ael-test/ael-test18/extensions.ael, + pbx/ael/ael-test/ael-test19/extensions.ael (added), + pbx/ael/ael-test/ael-test19 (added), + pbx/ael/ael-test/ref.ael-test20 (added), + pbx/ael/ael-test/ael-test20/extensions.ael (added), + pbx/ael/ael-test/ael-test20 (added): updated the ael regressions + to match what's in trunk + +2007-05-03 14:36 +0000 [r62912] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged + revisions 61357,61770,62885 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | + 1 line some fixes for PMP Hold/Retrieve, it should work now, when + briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 + (Di, 24 Apr 2007) | 1 line added lock for sending messages to + avoid double sending. shuffled some empty_chans after the + cb_event calls, this avoids that a release_complete from a quite + different call releases a fresh created setup by accident. + ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 + Mai 2007) | 1 line fixed the problem that misdn_write did not + return -1 when called with 0 samples in a frame this resultet in + a deadlock in some circumstances, when the call ended because of + a busy extension. added encoding of keypad. ........ + +2007-05-03 13:54 +0000 [r62883] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test18 (added), + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ael-test18/extensions.ael (added), + pbx/ael/ael-test/ael-test18 (added), + pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c, + pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7: + These mods fix bug 9623, where an '@' in the eswitch contents + causes a syntax error. I also updated the regressions. + +2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming + + * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 + May 2007) | 2 lines doh... initializing the pointer variable will + work just a bit better ........ + + * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 + May 2007) | 7 lines increase reliability and efficiency of static + Realtime config loading via ODBC: don't request fields we aren't + going to use don't request sorting on fields that are pointless + to sort on explicitly request the fields we want, because we + can't expect the database to always return them in the order they + were created (reported by blitzrage in person (!), patch by me) + ........ + + * res/res_config_pgsql.c: improve static Realtime config loading + from PostgreSQL: don't request sorting on fields that are + pointless to sort on use ast_build_string() instead of snprintf() + don't request the list of fieldnames that resulted from the query + when we both knew what they were before we ran the query _AND_ we + aren't going to do anything with them anyway (patch by me, + inspired by blitzrage's bug report about res_config_odbc) + +2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant + + * main/channel.c: Merge changes from team/russell/inband_dtmf ... + Fix some issues related to generating inband DTMF. There are two + changes here: 1) The list of DTMF tones in the senddigit_begin() + function explicitly specified 100ms of the tone followed by 100ms + of silence. This really broke things with the way that Asterisk + now wants complete control over when the digit begins and ends. + So, regardless of what Asterisk really wanted to do, this was + going to play out the tone at the length it wanted to. This + caused various problems like DTMF translation to inband to be + extremely unreliable. The list of tones has been changed so that + the correct DTMF tone is played indefinitely until Asterisk tells + it to stop. 2) ast_write() had to be modified to let a DTMF_END + frame get processed even when a generator is present. This is how + the tone will finally get stopped. (issues #8944, #9250, #9348, + maybe others. Thanks to mdu113 from #8944 for the testing and + feedback!) + + * main/manager.c: Backport the change that only went in to trunk + that fixes the command manager action over http. (reported + internally by pari and bkruse) + +2007-05-02 20:46 +0000 [r62738] Steve Murphy + + * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May + 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being + in 'h' extension louses up the dst field ........ + +2007-05-02 17:43 +0000 [r62692] Tilghman Lesher + + * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) + | 4 lines Issue 9638 - if a text frame is sent with no + terminating NULL through a bridged IAX connection, the remote end + will receive garbage characters tacked onto the end. ........ + +2007-05-02 17:10 +0000 [r62689] Steve Murphy + + * configs/extensions.conf.sample, main/channel.c, main/pbx.c, + channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the + clid, src fields in channel_alloc call. b)in the channel_alloc + func, set the cid_num and name fields from the arglist[blush]. c) + don't update the channel app & app data fields if you are in the + 'h' extension. d)the load_module func in cdr_radius needs to + return DECLINE, SUCCESS. + +2007-05-02 06:15 +0000 [r62624] Olle Johansson + + * channels/chan_sip.c: Don't unlock a channel that we already know + does not exist (propably isue 8228) + +2007-05-01 21:57 +0000 [r62548] Russell Bryant + + * /, res/res_features.c: Merged revisions 62547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | + 4 lines Remove an unnecessary check that makes it so if you hang + up after doing an attended transfer before the target extension + answers the channel, the transfer is not successful. (issue + #9338, patch by svanlund) ........ + +2007-05-01 21:34 +0000 [r62545] Tilghman Lesher + + * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user() + (found by rayjay, different fixes by me) + +2007-05-01 16:26 +0000 [r62497] Russell Bryant + + * /, configs/indications.conf.sample: Merged revisions 62496 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | + 3 lines Add indications.conf information for the Philippines. + (issue #9525, reported and patched by loloski) ........ + +2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | + 4 lines This patch fixes an issue where depending on the cause + code, when the network sends a PRI disconnect, the call may not + be properly hung up. (issue #9588, reported and patched by + softins) ........ + + * include/asterisk/http.h, main/http.c: When serving dynamic + content, include a Cache-Control header to instruct the browsers + to not store the resulting content. (issue #9621, reported by + Pari, patch by me) + +2007-04-30 14:52 +0000 [r62371] Jason Parker + + * configs/iax.conf.sample: Remove unused (and potentially + confusing) jitterbuffer options from sample config. + +2007-04-30 14:36 +0000 [r62369] Joshua Colp + + * main/asterisk.c, /: Merged revisions 62368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 + lines Update copyright notice. It's now the year 2007! ........ + +2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant + + * channels/chan_zap.c: Fix a bug that made the "language" setting + in zapata.conf not functional. (issue #9626, reported and fixed + by sergee) + + * apps/app_meetme.c: Note that the "talker optimization" option + will be enabled by default in 1.6 + +2007-04-27 Russell Bryant + + * Asterisk 1.4.4 released. + +2007-04-27 21:10 +0000 [r62218] Russell Bryant + + * channels/chan_agent.c: Fix a weird problem where when a caller + talking to someone sitting behind an agent channel sent a digit, + the digit would be played to the agent for forever. This is + because chan_agent always returned -1 from its send_digit_begin + and _end callbacks. This non-zero return value indicates to the + Asterisk core that it would like an inband DTMF generator put on + the channel. However, this is the wrong thing to do. It should + *always* return 0, instead. When the digit begin and end + functions are called on the proxied channel, the underlying + channel will indicate whether inband DTMF is needed or not, and + the generator will be put on that one, and not the Agent channel. + (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed + by me) + +2007-04-27 16:17 +0000 [r62174] Jason Parker + + * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 + lines This transcoder message needn't be a NOTICE. I've seen it + cause confusion more than a few times. ........ + +2007-04-27 16:14 +0000 [r62171] Russell Bryant + + * main/pbx.c: If no variables were passed into + pbx_substitute_variables_helper_full(), then don't even bother + creating a temporary bogus channel, since that is only for + allowing certain functions to operate on the variables as if they + were on a channel. Most importantly, this fixes a crash. (issue + #9613, reported by callguy, fixed by me) + +2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 + lines Issue #7351 - SIP Cancel fails due to the wrong contact + uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka + - THANKS!!!! THis was a hard one to catch. ........ + + * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some + annoying DEBUG messages not controlled by option_debug (DEA). + Thanks! + +2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp + + * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 + lines Revert previous fix for when the IAX2 channel goes funky + (that's the technical term). This is causing legit calls to be + prematurely hung up. (issue #9600 reported by justdave) ........ + + * main/channel.c: Missed an ast_app_group_discard during merge. + Thanks blitzrage! + + * res/res_monitor.c: Don't always say that the channel is being + paused if it is actually being unpaused in the Manager ack + message. (reported by jsmith in #asterisk-bugs) + + * main/config.c, /: Merged revisions 61958 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 + lines Don't count failed include attempts against the + configuration include level. (issue #9593 reported by mostyn) + ........ + +2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming + + * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) + | 2 lines handle a very bizarre race condition with channels + being redirected before a simple switch can be started on them + (issue #9286) ........ + +2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | + 2 lines If the callerid= option is specified, but empty, clear + any previous data. ........ + + * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | + 2 lines Ensure that callerid settings are reset on a reload. + ........ + +2007-04-25 19:21 +0000 [r61805] Joshua Colp + + * main/cli.c, main/channel.c, include/asterisk/app.h, + funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 + lines Merge rewritten group counting support. No more storing + data on the variable list of the channels. That was bad, mmmk? + (issue #7497 reported by sabbathbh) ........ + +2007-04-25 16:22 +0000 [r61799] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | + 3 lines Fix a typo where cid_num got copied instead of cid_ani. + (issue #9587, reported and patched by xrg) ........ + +2007-04-24 Russell Bryant + + * Asterisk 1.4.3 released. + +2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant + + * main/manager.c, /: Merged revisions 61786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | + 4 lines Don't crash if a manager connection provides a username + that exists in manager.conf but does not have a password, and + also requests MD5 authentication. (ASA-2007-012) ........ + + * main/channel.c, include/asterisk/channel.h: Improve DTMF handling + in ast_read() even more in response to a discussion on the + asterisk-dev mailing list. I changed the enforced minimum length + of a digit from 100ms to 80ms. Furthermore, I made it now enforce + a gap of 45ms in between digits. These values are not + configurable in a configuration file right now, but they can be + easily changed near the top of main/channel.c. + +2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard + + * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) + | 1 line removed #if 0 block from chan_phone, chan_zap, and + chan_modem restart_monitor() ........ + +2007-04-24 16:16 +0000 [r61774] Russell Bryant + + * main/dial.c: Add a few more state changes in + handle_frame_ownerless() so that the SLA code will get notified + of these changes even when an owner channel is not provided. This + isn't from a specific bug report, it's just something I noticed + while poking around. + +2007-04-24 16:07 +0000 [r61772] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 + lines Allow RFC2833 to be sent in the response SDP when an INVITE + comes in without SDP. (issue #9546 reported by mcrawford) + ........ + +2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant + + * main/pbx.c: Some dialplan functions, such as CUT(), expect to + operate on variables on a channel. So, this little hack lets them + work in places where a channel doesn't exist, such as within + DUNDi configuration. (issue #9465, reported and patched by + Corydon76, testing by blitzrage) + + * main/channel.c: Ensure that digits passing through Asterisk have + a reasonable minimum length. It is currently 100 ms. If someone + thinks this should be different, feel free to speak up. (related + to issues #8944, #9250, and #9348) + +2007-04-20 21:35 +0000 [r61705-61707] Jason Parker + + * main/rtp.c: Avoid invalid seqno cycling detection. Per comment + from Dave Troy: This adds back in some simple typecasting I had + in an earlier version which I realize now may be breaking things. + Issue #9554. + + * main/loader.c, /: Merged revisions 61704 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 + lines Fix an issue that I noticed while looking over issue 9571. + The reload timestamp was getting set after reloading the built-in + stuff, and before the modules. ........ + +2007-04-20 20:42 +0000 [r61697] Russell Bryant + + * main/rtp.c: Remove a stray debug message introduced by a recent + commit. + +2007-04-20 19:51 +0000 [r61694] Jason Parker + + * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 + lines If the '* to hangup' option is not enabled, we don't need + to disable * as a valid exit key. If it was enabled, this + statement would've never been checked in the first place. Issue + #9552 ........ + +2007-04-20 18:19 +0000 [r61690] Russell Bryant + + * main/config.c, apps/app_voicemail.c, main/manager.c, + include/asterisk/config.h: Fix the UpdateConfig manager action to + properly treat "variables" and "objects" differently (a=b versus + a=>b). (issue #9568, reported by pari, patch by me) + +2007-04-19 08:37 +0000 [r61686] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 + lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by + Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........ + +2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher + + * main/manager.c: Bug 9557 - simple reason why reading a function + always returned NULL + + * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c, + funcs/func_groupcount.c, /, funcs/func_timeout.c, + funcs/func_cdr.c: Merged revisions 61680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) + | 5 lines Bug 9557 - Specifying the GetVar AMI action without a + Channel parameter can cause Asterisk to crash. The reason this + needs to be fixed in the functions instead of in AMI is because + Channel can legitimately be NULL, such as when retrieving global + variables. ........ + +2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming + + * sounds/Makefile: allow external build systems to extract the + required sound file versions + +2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson + + * main/rtp.c: Clean upp formatting, add some doxygen stuff while + we're in cleaning mode... Thanks Kevin! + + * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy) + +2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson + + * channels/chan_sip.c: #9483, half of patch by twilson to solve 302 + redirect issues + + * /: Blocking AstHoloPatch from 1.2 + +2007-04-13 21:17 +0000 [r61658] Steve Murphy + + * main/cdr.c: This is a fix to the way CDR merge handles the data + that results from ForkCDR. + +2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 + lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves + the same as OUTBOUND_GROUP except it will get unset after use so + it won't get accidentally inherited. (issue #BE-140) ........ + + * apps/app_speech_utils.c: Do not bother looking for a result if + none are present. + + * channels/chan_sip.c: For those very verbose SIP implementations + that attach tons of info to the Contact header... let's increase + our variable sizes. (issue #9535 reported by jeffg) + +2007-04-13 17:10 +0000 [r61645] Russell Bryant + + * apps/app_voicemail.c: Eliminate a compiler warning with + ODBC_STORAGE enabled so that it will build under dev-mode. + +2007-04-13 17:01 +0000 [r61644] Steve Murphy + + * channels/chan_oss.c: A fix for chan_oss that resulted from the + CDR changes; it helps to use the right info. + +2007-04-13 16:32 +0000 [r61641] Joshua Colp + + * channels/chan_sip.c: Don't assume the callid of a dialog will be + set, as in some circumstances it may not. (issue #9534 reported + by tecnoxarxa) + +2007-04-11 16:05 +0000 [r61477] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | + 5 lines If someone sets the "useragent" option in sip.conf to be + empty, then don't add the User-Agent header at all. It is an + optional header, anyway. Also, the bug report says that some of + Japan's SIP providers don't allow it for some weird reason. + (issue #9488, reported by makoto, fixed by me) ........ + +2007-04-11 15:39 +0000 [r61443] Nadi Sarrar + + * channels/chan_misdn.c: Don't export AOCD variables on + misdn_hangup anymore, this was mainly a fix for trunk.. + +2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | + 6 lines Fix a bug with switching between host=dynamic and using + specific hosts for peers. The code would only reset the peer's + address when it is dynamic if it was a new peer structure. Now, + it will also reset the address if it was already in the peer + list, but before the reload, it was not dynamic. (issue #9515, + reported by caio1982, fixed by me) ........ + + * main/http.c: Add "svgz" to the mimetypes table. (issue #9510, + bkruse) In passing, constify the elements of the mimetypes table. + + * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | + 5 lines Remove the attempt at reporting configuration errors in + sip.conf. This can cause a bunch of improper messages when using + realtime. I give up. As oej tried to convince me when I put this + in, there is just no easy way to do it. (inspired by a message on + the -dev list) ........ + +2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar + + * channels/chan_misdn.c: Export AOCD variables on misdn_hangup. + + * channels/chan_misdn.c: Ignore facility messages in case we don't + have a corresponding channel object. + + * channels/chan_misdn.c: AOCD's are now exported to asterisk + channel variables. + +2007-04-10 16:05 +0000 [r61220] Russell Bryant + + * main/Makefile, main/http.c, main/minimime (removed): File upload + support was added to solve some needs for the Asterisk GUI. + However, after much discussion, it has been decided that adding + this to 1.4 is not in the best interests of the project. It has + been removed here, but will remain in trunk. + +2007-04-10 12:43 +0000 [r61183] Nadi Sarrar + + * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr + 2007) | 2 lines msns config parameter defaults to '*' ........ + +2007-04-10 05:18 +0000 [r61136] Steve Murphy + + * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a + previous fix to overcome a compiler warning; the app NoCDR() has + been updated to mark the channel CDR as POST_DISABLED instead of + destroying the CDR; this way its flags are propagated thru a + bridge and the CDR is actually dropped. The cases where only one + channel in a bridge has a CDR was cleaned up. + +2007-04-09 19:58 +0000 [r61072] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 + lines - Don't send ActionID before Response: header. - Don't use + a blank in an AMI header ........ + +2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming + + * main/minimime/mm_envelope.c, res/res_features.c: fix up some + warnings found using --enable-dev-mode + + * main/minimime/Doxyfile (removed), + main/minimime/tests/messages/CVS (removed), + main/minimime/tests/CVS (removed): remove some more stuff we + don't need + +2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant + + * main/minimime/test (removed): Remove another directory that + should no longer be there + + * main/minimime/Make.conf (removed), main/minimime/mytest_files + (removed), main/minimime/.cvsignore (removed), main/minimime/sys + (removed), main/minimime/mm-docs (removed): Remove various files + that I thought I already removed. + +2007-04-09 19:05 +0000 [r61022] Jason Parker + + * apps/app_queue.c: Use the appropriate interface name with + COMPLETECALLER. Issue 9395. + +2007-04-09 18:32 +0000 [r60989] Steve Murphy + + * channels/chan_oss.c, main/channel.c, main/cdr.c, + channels/chan_phone.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_features.c, + channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, + channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, + channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, + include/asterisk/channel.h, channels/chan_gtalk.c, + channels/chan_iax2.c: This is a big improvement over the current + CDR fixes. It may still need refinement, but this won't have as + many folks bothered. + +2007-04-09 18:02 +0000 [r60984] Olle Johansson + + * res/res_jabber.c: Add final new line after JabberEvent + +2007-04-09 17:22 +0000 [r60936] Jason Parker + + * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 + lines Allow matching on names shorter than 3 chars. This also + fixes the case where somebody wants to match on less then 3 + chars. Issue 9071 ........ + +2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher + + * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) + | 2 lines Don't check for error when lowering priority (according + to the manpage, it should never happen anyway). It might could + happen, though, if another thread messed with the priority, so + safeguard against that (reported via -dev list). ........ + + * channels/chan_local.c, /: Merged revisions 60846 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 + Apr 2007) | 2 lines Bug 9505 - If the return value for + local_queue_frame is set, then p->lock is no longer valid. + ........ + +2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 + lines When calling a device that then forwards us elsewhere... we + have to make our channels compatible if it is the only channel + being dialed. (issue #9445 reported by marcelbarbulescu) ........ + + * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if + MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) + +2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) + | 2 lines Gosub called within a Macro resets the arguments + improperly and causes general weirdness. (Issue 8329) ........ + + * main/http.c: Fix --enable-dev-mode + + * channels/chan_oss.c: Off by one error, resulting in a crash + (Issue 9500) + + * /, main/file.c: Merged revisions 60660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) + | 2 lines Bug 9486 - memory leak when opening a filestream + ........ + +2007-04-06 20:58 +0000 [r60603] Russell Bryant + + * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile, + main/minimime/mimeparser.yy.c, main/minimime/minimime.c, + main/manager.c, main/minimime/mm_mimepart.c, + main/minimime/test.sh, configure, include/asterisk/compat.h, + main/strcompat.c, main/minimime/mm_internal.h, main/http.c, + main/minimime/tests/parse.c, main/minimime/mm_base64.c, + main/minimime/mm_mimeutil.c, main/minimime/mm.h, + main/minimime/tests, main/minimime/mm_header.c, + main/minimime/mm_error.c, main/Makefile, + main/minimime/mm_codecs.c, main/minimime/mm_param.c, + configure.ac, main/minimime/Makefile, main/minimime/mm_init.c, + include/asterisk/manager.h, main/minimime/strlcpy.c, + configs/http.conf.sample, main/minimime/mm_parse.c, + main/minimime/tests/create.c, main/minimime/mm_contenttype.c, + main/minimime/mm_util.c, main/minimime/mm_envelope.c, + main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c, + main/minimime/tests/messages/test2.txt, + main/minimime/tests/messages/test3.txt, + main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, + main/minimime/tests/messages/test4.txt, + main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h, + main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c, + main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt, + main/minimime/mimeparser.l, main/minimime/mm_context.c, + main/minimime/mimeparser.tab.h, main/minimime (added), + main/minimime/mm_warnings.c, main/minimime/mm_queue.h, + main/minimime/tests/messages, include/asterisk/autoconfig.h.in, + main/minimime/mimeparser.y, Makefile.moddir_rules, + main/minimime/sys, main/minimime/tests/Makefile: To be able to + achieve the things that we would like to achieve with the + Asterisk GUI project, we need a fully functional HTTP interface + with access to the Asterisk manager interface. One of the things + that was intended to be a part of this system, but was never + actually implemented, was the ability for the GUI to be able to + upload files to Asterisk. So, this commit adds this in the most + minimally invasive way that we could come up with. A lot of work + on minimime was done by Steve Murphy. He fixed a lot of bugs in + the parser, and updated it to be thread-safe. The ability to + check permissions of active manager sessions was added by Dwayne + Hubbard. Then, hacking this all together and do doing the + modifications necessary to the HTTP interface was done by me. + +2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard + + * UPGRADE.txt: clarified a sentence in the format_wav section + + * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and + plan to remove GAIN code from trunk + +2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant + + * apps/app_meetme.c: When a station picks up a trunk that was on + hold, make the hints reflect that nobody has the trunk on hold + anymore. + + * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459, + reported by francesco_r, fixed by me) * The original behavior was + that if one station put a call on hold, another one picked it up, + and then hung up, the code would still consider the call on hold + by the first station, so the trunk would not be hung up. However, + to better comply with what most people seem to expect it to + behave, it will now hang up the trunk. * Fix a problem with + "barge=no". This was only intended to prevent people from joining + calls that are in progress. However, it also prevented other + people from picking up a call that was on hold. This has been + fixed. * When there are no active stations on a trunk and it is + on hold, the code now indicates the HOLD and UNHOLD conditions to + the trunk channel. This allows music on hold to be played to the + trunk when it is on hold. + +2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson + + * channels/chan_zap.c: Make sure we check the faxdetect option + before doing fax processing + + * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 + lines There should only be one code path for doing DTMF + conditionals on channels. This fixes it. ........ + +2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming + + * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) + | 2 lines remove undocumented 'cardsmode' parameter and stop + searching for transcoders during reload() ........ + +2007-04-06 01:14 +0000 [r60361] Joshua Colp + + * res/res_speech.c, apps/app_speech_utils.c, + include/asterisk/speech.h: Add support for returning different + types of results (ie: NBest). + +2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard + + * formats/format_wav.c: modified default GAIN for issue 5823, + thanks jrwalliker + +2007-04-05 22:35 +0000 [r60323] Steve Murphy + + * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added + some clarification to the example configs for CDRs, on how to + select a backend. Also, made cdr-csv the default if you 'make + samples', and no other changes. + +2007-04-05 16:10 +0000 [r60268] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 + lines Just because we can't find the voicemail configuration + file, doesn't mean that the module failed to load. The user could + be using realtime. Issue #9473 ........ + +2007-04-05 15:47 +0000 [r60265] Russell Bryant + + * main/http.c: Add the MIME type for gif by request from Pari + +2007-04-05 12:55 +0000 [r60214] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 + lines Only unlock our pvt and net locks if we are actually going + to try to lock the owner again. (issue #9472 reported by zoa) + ........ + +2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant + + * main/manager.c, /: Merged revisions 60134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | + 6 lines It is valid to redirect channels via the manager + interface that are not in the UP state. Instead of checking for + that to prevent to ensure a dead channel doesn't get redirected, + just use the ast_check_hangup() API call. (issue #9457, reported + by Callmewind, patch by me) (related to issue #8977) ........ + + * channels/chan_sip.c: Add a Content-Length of 0 to the response + built by transmit_response_with_unsupported(). (issue #9454, + reported by makoto, fixed by me) + + * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | + 4 lines Fix the return value of handle_common_options() so that + it always properly indicates whether it handled the option or + not. (issue #9455, reported by Netview, fixed by me) ........ + + * apps/app_meetme.c: Fix a problem where if a trunk was hung up + while it was on hold, all of the hints would reflect the line + still on hold, even though it should reflect that it is back to + not in use. (issue #9459, reported by francesco_r, fixed by me) + +2007-04-03 19:40 +0000 [r59963] Joshua Colp + + * apps/app_speech_utils.c: Don't clash when a person both speaks + and uses DTMF. + +2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | + 4 lines Don't attempt to report configuration errors in + build_user(). oej pointed out that for a "friend" entry, this + won't work, because all user options are valid for peers, but not + the other way around. ........ + + * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | + 3 lines Make chan_sip report when it encounters an unknown + option. (issue #9440, reported by nightcrawler) ........ + + * /, main/app.c: Merged revisions 59886 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | + 5 lines When doing a built-in blind or attended transfer, restore + the ability to use '#' to terminate the number and immediately do + the transfer instead of having to dial the number and just wait + for the feature digit timeout. (issue #8366, xueliangliang) + ........ + + * Makefile: Ensure that menuselect gets executed in dependency + check mode every time you run make. + +2007-04-03 11:02 +0000 [r59804] Nadi Sarrar + + * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h: + Merged revisions 59788,59803 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 + lines Use the new sysfs way of mISDN 1.2 to check if a port is NT + or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, + 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ + +2007-04-03 07:20 +0000 [r59774] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: + Merged revisions 59623-59624,59639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | + 1 line we can now make 30 channels on a PRI (before we forgot + chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 + (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ + r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | + 1 line added option which allows us to accept incoming SETUP + Messages without automatically sending Proceeding or Setup + Acknowledge, this is useful with some broken switches and if you + want to Release incoming calls without previously having + acknowledged them. The new option is + noautorespond_on_setup=yes|no default is no, so we don't break + the existing behaviour ........ + +2007-04-02 18:58 +0000 [r59724] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 + lines Increase the maximum size for a string of mailboxes to + 1024. (issue #9270 reported by rtucker) ........ + +2007-04-02 17:31 +0000 [r59688] Steve Murphy + + * pbx/pbx_ael.c: continue in for-loop should go to the incrementer, + not the test. As per 9435, thanks to marcelbarbulescu + +2007-04-02 15:39 +0000 [r59654] Russell Bryant + + * main/netsock.c, /: Merged revisions 59608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | + 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. + This is needed by the patch that went in for issue 7874. + chan_iax2 needs to be able to create socket that is lisetning on + INADDR_ANY, but also be able to bind sockets to specific + addresses. (Thanks to Stevenson on the asterisk-dev mailing list + for explaining why this flag was needed.) ........ + +2007-03-30 22:50 +0000 [r59573] Jason Parker + + * configure, main/Makefile, acinclude.m4: Add linux-uclibc host + arch..."thingy". Sorry, I don't know what it's called... + +2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy + + * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, + include/asterisk/cdr.h: several changes via kpflemings review + + * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, + include/asterisk/cdr.h: These mods fix CDR issues from 8221, + 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated + from transfer situations. + + * configs/extensions.conf.sample: A small clarification to keep + bugs from being filed, and confusion from rising, if + clearglobalvars is set, and globals are set in the AEL file. + (9419) + +2007-03-29 17:43 +0000 [r59363] Russell Bryant + + * res/res_jabber.c: When building a response to a subscription, the + "from" must be the full Jabber ID. This fixes some problems where + jabber users are not able to add their Asterisk account to their + user list, since they are unable to get Asterisk to approve their + subscription. (issue #8210, reported by caspy, and verified by + bradtem) + +2007-03-29 17:38 +0000 [r59361] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 + lines Keep a global array of variables indicating whether certain + conference rooms are in use. This ensures that two people going + into a new dynamic conference when the 'e' option is set don't go + into the same conference room. (issue #8835 reported by eliel) + ........ + +2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant + + * main/rtp.c, /: Merged revisions 59357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | + 5 lines If an error occurs when reading from an RTP socket, and + the error code does not indicate that we should try again, then + return NULL instead of a "null frame". This will prevent Asterisk + from trying over and over again, and eventually causing the + system to crash. (issue #8285, john) ........ + + * channels/chan_iax2.c: When the IAX2 read callback gets called, + return NULL instead of a "null frame". This will cause Asterisk + to hangup the call instead of keep trying whatever it was doing. + Under normal conditions, this function would *never* be called. + However, the author of this patch says an error will occur that + will cause it to get called every 100 thousand calls or so. When + this does happen, it puts the channel in a loop that eventually + brings down the system. So, hangup up the call is certainly a + better alternative. (issue #8286, john) + + * Makefile: Export the GTK2 library and include information to sub + Makefiles. + +2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher + + * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) + | 3 lines Issue 9415 - No point to getting a diagnostic field if + we aren't doing anything with the information. (Plus, it tends to + crash the Postgres ODBC driver.) ........ + +2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher + + * res/res_odbc.c: Another crash that I thought we had fixed already + - Issue 9396 + + * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) + | 2 lines Oops ........ + + * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) + | 2 lines Fix a few remaining bad mmap(2) return values ........ + +2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant + + * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | + 3 lines Fix the check of the return value from mmap(). Thanks to + Corydon for catching this one. ........ + + * apps/app_directory.c: Fix app_directory to actually compile with + ODBC_STORAGE, and update the code to the latest res_odbc API. + + * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used. + The Makefile did not properly ensure that this information got + copied from what was selected for app_voicemail. (issue #9224) + + * channels/chan_sip.c: Fix the check that ensures that the CHANNEL + function's first argument is "rtpqos". Thanks, Corydon. :) + +2007-03-27 18:16 +0000 [r59261] Steve Murphy + + * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes + asterisk), kpfleming pointed on asterisk-dev, that DECLINE in + this case the proper thing to do. This change now has it doing + the proper thing. + +2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | + 4 lines Fix the use of the "sourceaddress" option when "bindaddr" + is set to 0.0.0.0 instead of having each interface explicitly + listed. (issue #7874, patch by stevens) ........ + + * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS + function to just be additional parameter of the CHANNEL function. + This way, it will be possible for other RTP based channel drivers + to expose this information in the future. + +2007-03-27 15:00 +0000 [r59254] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 + Mär 2007) | 1 line fixed #9355 ........ + +2007-03-26 21:45 +0000 [r59230] Tilghman Lesher + + * channels/chan_sip.c: Oops, this should be case insensitive + +2007-03-26 21:41 +0000 [r59228] Steve Murphy + + * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes + asterisk). I turned a duplicate context from a WARNING to an + ERROR. Now you get a module load failure, and asterisk just + exits. That's better than a crash, right\? + +2007-03-26 21:37 +0000 [r59227] Tilghman Lesher + + * channels/chan_sip.c: Change this to a single dp function to make + oej happy. + +2007-03-26 20:06 +0000 [r59225] Steve Murphy + + * main/config.c: Fix for 9257; by eliminating the globals in + main/config.c, we make it thread-safe, which is a minimum + requirement. + +2007-03-26 19:34 +0000 [r59223] Joshua Colp + + * apps/app_speech_utils.c: Add ability to specify no timeout. This + means as soon as the prompt is done playing it moves on to the + next priority. + +2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant + + * apps/app_voicemail.c: Somehow the code for building the email for + voicemail got out of sync. This change makes a few tweaks to get + 1.4 in sync with trunk. (issue #9301) + + * apps/app_meetme.c: Fix some codec negotiation problems when + CallerID support is not enabled in SLA. (issue #9308, reported by + twilson) + +2007-03-26 18:13 +0000 [r59213] Joshua Colp + + * apps/app_speech_utils.c: Make SpeechBackground obey the digit + timeout value. + +2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant + + * channels/chan_sip.c: Rename the new dialplan functions to match + the variable name + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The + AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in + some because they get set in sip_hangup. So, there are common + situations where the variables will not be available in the + dialplan at all. So, this patch provides an alternate method for + getting to this information by introducing AUDIORTPQOS and + VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, + with some testing by blitzrage) + +2007-03-26 17:38 +0000 [r59206] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, + pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE, + and STANDALONE_AEL + +2007-03-26 15:25 +0000 [r59202] Nadi Sarrar + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure, + include/asterisk/autoconfig.h.in, channels/misdn/Makefile, + channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2 + provides a dsp pipeline for i.e. echo cancellation modules, make + chan_misdn use it. * add a check for linux/mISDNdsp.h to + configure.ac and update the autogenerated files: 'configure', + 'autoconfig.h.in' (the 'configure' script was not in sync with + the latest configure.ac, so the diff is a bit bigger than + expected). + +2007-03-26 15:16 +0000 [r59200] Joshua Colp + + * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the + aelparse binary! DONT_OPTIMIZE should now work once again. + +2007-03-24 01:39 +0000 [r59195] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 + lines Only try to handle a response if it has a response code. + (ASA-2007-011) ........ + +2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy + + * /: blocking out the fix in 59187... already incorporated here + + * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 + line Added a few words in the Macro doc strings about the + behavior of macros with hangups (et al.), as per 9337 ........ + +2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming + + * channels/chan_sip.c: don't allow string input to overrun the + buffer to hold it (ASA-2007-010) + + * channels/chan_misdn.c: remove variables that are no longer used + (--enable-dev-mode is good, developers should be using it) + +2007-03-22 14:40 +0000 [r59145] Steve Murphy + + * utils/Makefile: The stuff in utils was compiling with -O6 even if + DONT_OPTIMIZE is set in menuconfig. Added the include to fix that + +2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp + + * main/http.c: Add svg mimetype for pari. + + * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 + lines Indicate the filename changed when it is changed. (issue + #9311 reported by jsmith) ........ + + * channels/chan_sip.c: Until we can do media level parsing for + sendrecv/etc just use the first value found. This crept up when a + phone was offered audio+video and returned an inactive video + stream. chan_sip thought the phone said to put the person on hold + but that was totally wrong. (issue #9319 reported by benbrown) + +2007-03-20 21:04 +0000 [r59078] Tilghman Lesher + + * main/logger.c: Fix defines for inline stack backtraces (only used + by developers anyway) + +2007-03-20 20:42 +0000 [r59076] Joshua Colp + + * channels/iax2-parser.c: Copy len variable as well, should fix + remaining IAX2 DTMF issues. + +2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy + + * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should + return it to its previous, untouched, state. + + * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h: + The fix for the AEL <> (bug 9316) is here... + +2007-03-20 13:16 +0000 [r59064] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h: Merged revisions + 58849-58850,59062-59063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | + 1 line added method standard_dec for dialing out on groups, to + avoid conflicts, which caused issues with some ISDN providers + ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 + Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | + crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line + avoid sending a disconnect when we already received one. ........ + r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | + 1 line modified a loglevel ........ + +2007-03-19 Jason Parker + + * Asterisk 1.4.2 released. + +2007-03-19 22:29 +0000 [r59049] Tilghman Lesher + + * funcs/func_strings.c: Oops, this should have been a %d all along + +2007-03-19 15:52 +0000 [r59042] Joshua Colp + + * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295 + reported by ajohnson) + +2007-03-19 15:42 +0000 [r59040] Tilghman Lesher + + * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported + via -dev list) + +2007-03-18 20:37 +0000 [r59037] Olle Johansson + + * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return + code 0 (reported by qwerty1979) + +2007-03-18 16:36 +0000 [r59035] BJ Weschke + + * apps/app_followme.c: Don't return a non-zero return code if the + profile doesn't exist, to match what the documentation says it + already does. (#9307 Reported by kkiely) + +2007-03-16 16:12 +0000 [r58992] Joshua Colp + + * apps/app_page.c: Wait for the async thread to exit when hanging + up all of the paged phones under all circumstances. (issue #9181 + reported by PhilSmith) + +2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant + + * configs/sla.conf.sample: fix a couple SLA documentation + references + + * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex + (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added), + doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added), + doc/channelvariables.txt (added), doc/ael.txt (added), + doc/billing.tex (removed), build_tools/prep_tarball, + doc/callingpres.txt (added), doc/enum.txt (added), + doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added), + doc/cdrdriver.tex (removed), build_tools/make_buildopts_h, + doc/security.txt (added), doc/imapstorage.txt (added), + doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed), + doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac, + doc/iax.txt (added), doc/ael.tex (removed), + doc/channelvariables.tex (removed), doc/enum.tex (removed), + doc/security.tex (removed), doc/math.txt (added), Makefile, + doc/imapstorage.tex (removed), doc/privacy.tex (removed), + doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt + (added), apps/app_voicemail.c, doc/cliprompt.txt (added), + doc/chaniax.txt (added), doc/app-sms.txt (added), + doc/ast_appdocs.tex (removed), doc/realtime.tex (removed), + doc/ices.txt (added), doc/dundi.tex (removed), + doc/linkedlists.txt (added), doc/queuelog.txt (added), + doc/extconfig.txt (added), doc/radius.txt (added), + doc/cliprompt.tex (removed), doc/chaniax.tex (removed), + doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex + (removed), doc/ices.tex (removed), doc/asterisk.tex (removed), + doc/queuelog.tex (removed), doc/configuration.txt (added), + doc/asterisk-conf.txt (added), doc/sla.pdf (added), + doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt + (added), doc/mp3.tex (removed), doc/configuration.tex (removed), + doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added), + doc/channels.txt (added), doc/ip-tos.tex (removed), + doc/extensions.txt (added), doc/queues-with-callback-members.txt + (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added), + doc/misdn.txt (added), doc/manager.txt (added), + doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), + doc/billing.txt (added), doc/localchannel.txt (added), + doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt + (added), doc/00README.1st (added): Making these documentation + changes in the 1.4 branch upset various people, so these chanes + will only be done in the trunk. + + * build_tools/prep_tarball: Add the --pdf option to the usage of + rubber in prep_tarball + + * Makefile, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + configure script checking for GTK2 and some additional Makefile + targets to support gmenuselect + +2007-03-15 23:52 +0000 [r58946] Tilghman Lesher + + * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match + common syntax and update the resulting appdocs TeX file + +2007-03-15 23:24 +0000 [r58941] Russell Bryant + + * doc/asterisk.tex: add a link to the rubber homepage + +2007-03-15 23:11 +0000 [r58939] Tilghman Lesher + + * apps/app_setcdruserfield.c, main/pbx.c, + apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c: + Expand deprecation warnings from simply warning on use to the + builtin documentation. + +2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant + + * doc/asterisk.tex, Makefile: Add Asterisk version information to + the generated PDF + + * build_tools/prep_tarball: have prep_tarball attempt to build + asterisk.pdf + +2007-03-15 22:32 +0000 [r58933] Tilghman Lesher + + * funcs/func_realtime.c: Function works fine, but the documentation + is backwards. + +2007-03-15 22:25 +0000 [r58931] Russell Bryant + + * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex + (added), doc/freetds.txt (removed), doc/odbcstorage.txt + (removed), configure, doc/sla.tex, doc/cygwin.txt (removed), + doc/model.txt (removed), doc/channelvariables.txt (removed), + doc/ael.txt (removed), doc/billing.tex (added), + doc/callingpres.txt (removed), doc/enum.txt (removed), + doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed), + doc/cdrdriver.tex (added), build_tools/make_buildopts_h, + doc/security.txt (removed), doc/imapstorage.txt (removed), + doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added), + doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac, + doc/iax.txt (removed), doc/ael.tex (added), + doc/channelvariables.tex (added), doc/enum.tex (added), + doc/security.tex (added), doc/math.txt (removed), Makefile, + doc/imapstorage.tex (added), doc/privacy.tex (added), + doc/realtime.txt (removed), doc/dundi.txt (removed), + doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt + (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), + doc/ast_appdocs.tex (added), doc/realtime.tex (added), + doc/ices.txt (removed), doc/dundi.tex (added), + doc/linkedlists.txt (removed), doc/queuelog.txt (removed), + doc/extconfig.txt (removed), doc/radius.txt (removed), + doc/cliprompt.tex (added), doc/chaniax.tex (added), + doc/hardware.txt (removed), doc/mp3.txt (removed), + doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex + (added), doc/queuelog.tex (added), doc/configuration.txt + (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf + (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), + doc/h323.txt (removed), doc/mp3.tex (added), + doc/configuration.tex (added), doc/asterisk-conf.tex (added), + doc/jitterbuffer.txt (removed), doc/channels.txt (removed), + doc/ip-tos.tex (added), doc/extensions.txt (removed), + doc/queues-with-callback-members.txt (removed), doc/apps.txt + (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt + (removed), doc/manager.txt (removed), doc/jitterbuffer.tex + (added), doc/extensions.tex (added), doc/billing.txt (removed), + doc/localchannel.txt (removed), + doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt + (removed), doc/00README.1st (removed): Merge changes from + svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc + directory into a single LaTeX formatted document so that we can + generate a PDF, HTML, or other formats from this information. * + Add a CLI command to dump the application documentation into + LaTeX format which will only be include if the configure script + is run with --enable-dev-mode. * The PDF turned out to be close + to 1 MB, so it is not included. However, you can simply run "make + asterisk.pdf" to generate it yourself. We may include it in + release tarballs or have automatically generated ones on the web + site, but that has yet to be decided. + +2007-03-15 18:13 +0000 [r58923] Joshua Colp + + * channels/chan_iax2.c: Don't assume that the pvt structure will + still exist after calling schedule_delivery as it may not. (issue + #9278 reported by fmachado) + +2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant + + * channels/chan_sip.c: Some people like to put "limitonpeer" + instead of "limitonpeers" in their configuration. While we're at + it, support "limitonpeerz" and "limitonpeerssssss". (inspired by + issue #9172) + + * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the + examples section + + * doc/security.txt, /: Merged revisions 58896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | + 3 lines Add a note to the security file that the Asterisk CLI and + log files may contain sensitive information, and that people + should keep this in mind. ........ + + * configs/sla.conf.sample, apps/app_meetme.c: By default, don't + attempt to do any CallerID handling at all with SLA because it is + known to not work properly in some situations. However, add an + option to enable it for those that would like to use it anyway. + The short story behind this is that to properly handle CallerID + with SLA, we need the ability to change the CallerID on an + existing call, and we are not ready to handle that. + +2007-03-14 01:47 +0000 [r58880] Tilghman Lesher + + * funcs/func_strings.c: Issue 9162 - + pbx_substitute_variables_helper assumes the buffer is initialized + to all zeroes. This fixes a case where it wasn't. + +2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant + + * apps/app_meetme.c: Ensure that the blinky lights show that the + trunk stopped ringing when the trunk hangs up before a station + has answered it. (issue #9234, reported by francesco_r) + + * configs/sla.conf.sample: fix the reference to the SLA + documentation + +2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 + lines Issue #9229 - No port in request URI on register to non + default SIP ports (neelakantan) ........ + + * channels/chan_sip.c: Don't hangup the call on OK or errors on + MESSAGE and INFO inside of a dialog (like video update requests). + + * channels/chan_sip.c: Issue #9251 - Clear From URI from user + attributes (tgrman) + +2007-03-12 13:08 +0000 [r58825-58826] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 57034,57523,57753,58558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | + 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com + bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 + 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ + r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | + 1 line fixed another place where the out_cause was hardcoded to + 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 + Mar 2007) | 1 line we can free channel 31 as well, since we can + occupy it ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: added UU transceiving and + corect handling for rdnis + +2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp + + * main/rtp.c: Allow RFC2833 compensation to compensate for even + stupider implementations by queueing up the end frame at the + start, not the actual end. (issue #8963 reported by AndrewZ) + + * channels/chan_sip.c, configs/sip.conf.sample: Add + matchexterniplocally setting which only substitutes your + externip/externhost setting if it matches the localnet setting. I + know of at least two people who need opposite settings, so I made + it an option! (issue #8821 reported by kokoskarokoska) + +2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant + + * channels/chan_iax2.c: Fix a few more places in chan_iax2 where + the ast_frame used for receiving a frame was not properly + initialized. - Interpolating a frame when the jitterbuffer is in + use - decrypting a frame when IAX2 encryption is on - frames in + an IAX2 trunk + + * apps/app_meetme.c: Make the compiler happy and initialize a + variable. + + * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added): + Merge some updates to the SLA documentation. I plan to keep + working on this to explain all of the expected behavior with call + handling, configuration details for specific phones, and other + things. However, I got tired of doing it in plain text, so I + switched to using LaTeX. I have included the PDF version. I + haven't been able to get a nice looking plain text version out of + it yet, but I'm not terribly concerned since this is supposed to + be more of the manual, while the plain text sample configuration + file is the reference. + +2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp + + * apps/app_voicemail.c: Fix spelling of unavailable in voicemail + documentation. (issue #9248 reported by tensai) + + * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 + lines If we are unable to lookup the host in a c line we have to + abort, otherwise the previous data is gone and we will + (potentially) have no data when all is said and done. ........ + +2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant + + * apps/app_meetme.c: Hang up the channel that put the call on hold + in the event processing thread to avoid a race condition. Also, + if the station originated the call that it is putting on hold, + don't hang up the trunk if it was the only station on the call + and it is hanging up due to hold and not a normal hangup. + + * channels/chan_zap.c: Add a missing break statement so that + handling the above event does not incorrectly destroy the + channel. (issue #9242, andrew) + +2007-03-08 21:33 +0000 [r58479] Tilghman Lesher + + * res/res_odbc.c: Fix segfault (Issue 9236) + +2007-03-08 20:54 +0000 [r58474] Russell Bryant + + * apps/app_meetme.c: Refactor hold handling a bit so that it does + not require keeping the call up when a call is put on hold. + +2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp + + * main/rtp.c: Make early SDP seeding even smarter! We have to check + codecs in the make_compatible function too. (issue #9221 reported + by marcelbarbulescu) + + * main/dsp.c, /: Merged revisions 58388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 + lines Only print out debug message if the definition that makes + the variables shows up was actually defined. (issue #9233 + reported by serginuez) ........ + +2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming + + * main/http.c: this change was not needed; fclose() handles closing + the file descriptor already + + * apps/app_meetme.c: fix a compiler warning, and overwriting 'res' + value + + * main/http.c: fix two cases where HTTP session file descriptors + would not be closed + +2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant + + * channels/chan_zap.c, configure, configure.ac: If we receive + ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, + tzafrir) Also, update the configure script to make sure that we + don't try to build chan_zap if the installed version of zaptel + does not include ZT_EVENT_REMOVED. + + * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey + Moore) Merged revisions 58242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | + 7 lines Fix a problem where the Asterisk channel name could be + that of the wrong IAX2 user for a call. This is because the first + step of choosing this name is to look for an IAX2 peer that + happens to have the same IP/port number that this call is coming + from and assuming that is it. However, this is not always + correct. So, I have made it change this name after authentication + happens since at that point, we have an exact match. ........ + +2007-03-07 17:52 +0000 [r58240] Joshua Colp + + * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have) + at least one matching codec before attempting early bridge SDP + seeding. (issue #9221 reported by marcelbarbulescu) + +2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant + + * main/manager.c, /: Merged revisions 58164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | + 4 lines If the channels acquired using the manager Redirect + action are not up, then don't attempt to do anything with them. + It could lead to weird behavior, including crashes. (issue #8977) + ........ + +2007-03-06 23:10 +0000 [r58121] Steve Murphy + + * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 + line Fix for 9220: Eyebeam cannot renew subscriptions for + presence info. Reason: re-SUBSCRIBE requests don't include Accept + headers, which the rfc says are optional (to put it tersely), (it + uses MAY), and luckily, the sip_pvt struct has the format info + stored, so we simply leave it if the format is set, and the + accept header null. ........ + +2007-03-06 23:00 +0000 [r58119] Russell Bryant + + * configs/voicemail.conf.sample: Clarify the documentation of the + dialout and sendvoicemail options. (issue #9000, caio1982 and + serge-v) + +2007-03-06 20:37 +0000 [r58053] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 + lines Change error message to proper message ........ + +2007-03-06 18:01 +0000 [r58023] Russell Bryant + + * channels/chan_skinny.c: Return an error of transmit_response is + called without a session. (issue #9002) + +2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp + + * channels/chan_iax2.c: Since chan_iax2 does not support reception + of DTMF with duration ensure that it is set to 0 on the frame. + (issue #8521 reported by gdhgdh) + + * apps/app_meetme.c: Don't create a listen channel and record the + conference unless the option is turned on. (issue #9204 reported + by francesco_r) + + * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 + lines Make create_dirpath use our standard for return values. -1 + is failure, 0 is success. (issue #9205 reported by ballares) + ........ + +2007-03-05 15:20 +0000 [r57826] Steve Murphy + + * main/pbx.c, /: Merged revisions 57825 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 + line Fixed a typo introduced via 9156 (either the gotos or their + doc strings are wrong) ........ + +2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp + + * main/slinfactory.c: Don't allow a NULL pointer to reach + ast_frdup. (issue #9155 reported by cmaj) + + * res/res_jabber.c: Don't reference a potentially NULL pointer. + (issue #9199 reported by klolik) + + * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198 + reported by edgreenberg) + +2007-03-03 15:31 +0000 [r57707] Steve Murphy + + * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7: + Updated the regression tests + +2007-03-03 06:45 +0000 [r57649] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) + | 2 lines Memory leak of a list, if call recording was abandoned + ........ + +2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard + + * main/say.c: submitted patch for Georgian language, issue 9010, + submitted by Alexander Shaduri + +2007-03-03 00:02 +0000 [r57591] Russell Bryant + + * configs/sla.conf.sample: add missing configuration template. + Thanks to Lacy Moore on asterisk-users for pointing this out\! + +2007-03-02 Russell Bryant + + * Asterisk 1.4.1 released. + +2007-03-02 23:03 +0000 [r57556] Russell Bryant + + * configure, configure.ac: Update the check that is used to + determine whether zaptel transcoder support is present. The + interface has changed. + +2007-03-02 17:06 +0000 [r57477] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 + lines If a SIP message comes in and goes to a method handler that + requires additional values that may not be present then send back + an error. ........ + +2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy + + * main/pbx.c, /: Merged revisions 57458 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 + line further refinement in wording of goto documentation, as per + 9156, goto not proceeding to next instruction ........ + + * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes + right, but 9184 points out the problem-- the escape is removed by + pbx_config, and pbx_ael should also, before sending it down into + the pbx engine. Also, you have to insert it back in, if you are + generating extensions.conf code from the AEL. + +2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant + + * main/file.c: Return the correct digit that interrupted the + stream. This fixes exiting the Background application when using + the m option. (issue #9176, mjagdis) + + * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, + include/asterisk/channel.h: Merge changes from + svn/asterisk/team/russell/sla_updates * Originally, I put in the + documentation that only Zap interfaces would be supported on the + trunk side. However, after a discussion with Qwell, we came up + with a way to make IP trunks work as well, using some things + already in Asterisk. So, here it is, this now officially supports + IP trunks. * Update the SLA documentation to reflect how to setup + IP trunks. * Add a section in sla.txt that describes how to set + up an SLA system with voicemail. * Simplify the way DTMF + passthrough is handled in MeetMe. * Fix a bug that exposed itself + when using a Local channel on the trunk side in SLA. The + station's channel needs to be passed to the dial API when dialing + the trunk. * Change a WARNING message to DEBUG in channel.h. This + message is of no use to users. + +2007-03-01 22:21 +0000 [r57318] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 57317 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar + 2007) | 2 lines Don't even attempt to optimize things when a + proxy channel is involved. It will just explode in weird and + unexplaineable ways. (issue #9175 reported by + clegall_proformatique) ........ + +2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development + + * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. + +2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant + + * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla + docs + + * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes + from svn/asterisk/team/russell/sla_updates * Add support for + private hold. By setting "hold=private" for a trunk, only the + station that put the call on hold will be able to retrieve it + from hold. Also, by setting "hold=private" for a station, any + call that station puts on hold can only be retrieved by that + station. + + * apps/app_meetme.c: Minor formatting change + + * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from + svn/asterisk/team/russell/sla_updates * Add support for the + "barge=no" option for trunks. If this option is set, then + stations will not be able to join in on a call that is on + progress on this trunk. + +2007-02-28 19:23 +0000 [r57139] Steve Murphy + + * main/pbx.c, /: Merged revisions 57118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 + line a small documentation update, to reflect reality in the goto + doc strings, as per 9156, Goto does not proceed to next prio if + jump fails ........ + +2007-02-28 18:57 +0000 [r57093] Joshua Colp + + * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb + 2007) | 2 lines Fix a few more issues with the agent logoff CLI + command. (issue #9123 reported by arbrandes) ........ + +2007-02-28 18:20 +0000 [r57089] Russell Bryant + + * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of + changes from svn/asterisk/team/russell/sla_updates * Add support + for station ring delays. Ring delays can be set globally for a + station or for specific trunks on the station. * Fix a few bugs + in existing code. * Restructure and Reorganize code to improve + readability and maintainability. * Improve formatting of the "sla + show (trunks|stations)" CLI commands. + +2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp + + * apps/app_meetme.c: Picky compiler... + + * apps/app_speech_utils.c: Better handle timeouts when the + individual speaks after everything has been played but before the + timeout ends. + +2007-02-28 17:15 +0000 [r57049] Steve Murphy + + * pbx/pbx_ael.c: I was surprised that I had not yet downgraded + missing goto targets and macro call defs to a warning, in case + they are in extensions.conf; I rectified this problem. Also, A + goto in a macro to a target in a catch block was not being found; + I fixed this too; the cause was that I needed to treat catch + statements like an extension in the find_match code. + +2007-02-27 17:36 +0000 [r56975] Russell Bryant + + * apps/app_voicemail.c: Fix voicemail email attachments. I missed + the conversion of one of the line endings and there was an extra + one where it should not have been. (issue #9128) + +2007-02-26 22:01 +0000 [r56922] Tilghman Lesher + + * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, + picky... show deprecation warning in application help, too + (reported via list) + +2007-02-26 20:42 +0000 [r56888] Russell Bryant + + * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where + if a device was not specified in alsa.conf, then we just use the + system default, instead of creating our own default of hw:0,0. + (issue #9139) + +2007-02-26 20:07 +0000 [r56856] Joshua Colp + + * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 + lines Obey the clearglobalvars option in extensions reload (or + dialplan reload depending on your version). (issue #9146 reported + by ramonpeek) ........ + +2007-02-26 20:04 +0000 [r56847] Russell Bryant + + * channels/chan_iax2.c: Fix a crash in my last change to + iax2_indicate(). (issue #9150) + +2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp + + * apps/app_record.c: Update app_record documentation to use new CLI + command, core show file formats. (issue #9151 reported by junky) + + * main/pbx.c: Use ast_strlen_zero to see if the language and/or + context argument is not present for Background instead of just + checking if it is NULL. (issue #9141 reported by mjagdis) + +2007-02-26 16:51 +0000 [r56785] Russell Bryant + + * channels/chan_iax2.c: Do more complete locking of the + chan_iax2_pvt struct in the indicate callback. (Problem brought + up by Ben Smithurst on the asterisk-dev list) + +2007-02-26 16:36 +0000 [r56783] Joshua Colp + + * main/asterisk.c: Allow both of the show version files and core + show file versions CLI commands to work. (issue #9135 reported by + mvanbaak) + +2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant + + * apps/app_meetme.c: Move a comment to be in the correct struct. + +2007-02-25 14:46 +0000 [r56685] Tilghman Lesher + + * main/channel.c, /: Merged revisions 56684 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) + | 3 lines Issue 9130 - If prev is the last item on the channel + list, then evaluating additional conditions (e.g. name prefix) + will cause a NULL dereference. ........ + +2007-02-24 02:02 +0000 [r56569] Jason Parker + + * channels/chan_skinny.c: Make sure to set a speeddials parent on + creation. Don't crash if hold is pressed when no call is active. + Don't return in places that we shouldn't.. + +2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming + + * codecs/codec_zap.c: update to match zaptel 1.4 API change that + was committed a few minutes ago + +2007-02-23 23:24 +0000 [r56505] Russell Bryant + + * main/asterisk.c, /: Merged revisions 56504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | + 8 lines Fix up a couple more signal handlers to not do bad things + that could cause various undesirable results. The other day, I + made Asterisk deadlock by hitting Control-C because of a bad + signal handler. Now, signal handlers just set a flag and write to + an alert pipe for the flag to be handled. Then, there is another + thread that is monitoring for these flags. If being run in + console mode, it is just the main thread. If Asterisk is in the + background, a thread is created to do it. ........ + +2007-02-23 21:53 +0000 [r56457] Joshua Colp + + * main/sched.c: Change log notice to debug. It is possible for a + scheduled item to execute and be deleted at close to the same + time and unavoidable. If this happens this message creeps up. + +2007-02-23 20:20 +0000 [r56407] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | + 4 lines Don't destroy mutexes before unregistering all of the + entry points from the core. Also, fix a potential memory leak + from not destroying the locks for all of the possible call + numbers (about 32k of them). ........ + +2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming + + * build_tools/make_version_h: build special version strings for + AADK/S800i builds + +2007-02-23 17:58 +0000 [r56341] Russell Bryant + + * apps/app_voicemail.c: The IMAP storage code uses the same code to + build the email that is used when voicemail is sent via email + using something like sendmail. In the patch from bug 8033 to fix + various IMAP storage problems, the line endings in the email file + were changed in the code from "\n" to "\r\n". However, this + breaks sending regular voicemail to email. So, this change + conditionally sets line endings to "\r\n" only if IMAP_STORAGE is + enabled. (issue #9128, patch by jarjarbinks, modified by me to + not break IMAP storage) + +2007-02-22 23:08 +0000 [r56277] Russell Bryant + + * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, + doc/sla.txt: Merge changes from team/russell/sla_updates. This + batch of changes to the SLA code does a few different things. * I + made the SLA code event driven instead of having to act in a lot + of busy loops while dialing things to wait for state changes. + This makes the code more efficient and readable at the same time. + * I have implemented a couple of new features. The first is + inbound trunk ringing timeouts. This is an option that defines + how long to let an incoming call on a trunk to ring. * I have + also implemented ring timeouts for stations. They may be + specified for the entire station, meaning it is how long to let + the station ring before giving up. You can also specify a ring + timeout for a specific trunk on a station. So, you can say that + you only want a specific station to ring 5 seconds if it is line1 + ringing, but otherwise, there is no timeout. + +2007-02-22 18:49 +0000 [r56231] Joshua Colp + + * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 + lines Only change the original or clone channel if it's the + channel behind the proxy channel, not if it's just a regular + bridged channel. ........ + +2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development + + * doc/osp.txt: Update OSP documentation for v1.4. + +2007-02-22 10:33 +0000 [r56125] Olle Johansson + + * channels/chan_sip.c: Move message from verbose to debug + +2007-02-22 02:39 +0000 [r56094] Steve Murphy + + * sounds/Makefile: updated the sound tarball versions in Makefile + +2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant + + * channels/chan_sip.c: Restructure a little bit of code to reduce + nesting. There is no functionality change here. + + * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | + 3 lines If we receive a frame that is not in any of the + negotiated formats, then drop it. (potentially issue #8781 and + SPD-12) ........ + +2007-02-22 00:35 +0000 [r56008] Joshua Colp + + * main/cli.c: Print out deprecation notice on usage output of CLI + commands. (issue #8925 reported by blitzrage) + +2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming + + * main/loader.c: disable unloading of embedded modules... there is + a fundamental problem with doing so that will not be fixed in + this version of Asterisk due to its invasiveness + +2007-02-21 20:35 +0000 [r55957] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 + lines Change naughty warning message to provide useful + information. If a write now fails on a channel in meetme it will + tell you the channel name instead of spitting out the wrong error + message. ........ + +2007-02-21 20:27 +0000 [r55954] Jason Parker + + * channels/chan_gtalk.c: Fix locking issue, and accept + "transport-accept" as a valid accept message. This should solve + issues 8970 and 8503. + +2007-02-21 20:22 +0000 [r55951] Russell Bryant + + * apps/app_meetme.c: Simplify the last change to app_meetme, and + move the call to dispose_conf() up into the block where we know a + conf exists. + +2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp + + * apps/app_meetme.c: Only dispose of the conference if one was + created. + + * apps/app_speech_utils.c: Only start playing the next file if we + have not been quieted. + + * channels/chan_sip.c: Add a flag that indicates whether a SIP + dialog is an outgoing call or not. SIP_OUTGOING originally did it + but it was repurposed to the direction of the last transaction, + which can cause update_call_counter to falsely decrease the wrong + counters. (please don't hurt me oej) (issue #8943 reported by + mdu113) + +2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming + + * /, build_tools/make_version: Merged revisions 55868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 + Feb 2007) | 2 lines use new tag version script ........ + +2007-02-21 08:32 +0000 [r55834] Olle Johansson + + * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly + after transfer (decrement inuse early on transferer's call leg) + +2007-02-21 02:01 +0000 [r55799] Jason Parker + + * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. + Issue 7764, patch by sailer + +2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant + + * apps/app_meetme.c: Improve the reference counting to fix bugs + where people report seeing conferences listed that have no + members. (issue #9073) + +2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp + + * apps/app_voicemail.c: Better handle dropped IMAP connections. + (issue #9054 reported by bsmithurst) + + * channels/chan_sip.c: Return behavior I removed. I did not + remember that you could just add a localnet entry to make it + work. + + * channels/chan_sip.c: Don't test our own address against the + localnet settings. At least one person has had issues as a result + of this from #7051 so I'm reversing it. (issue #8821 reported by + kokoskarokoska) + + * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb + 2007) | 2 lines Defer clearing callback information if channels + are up until they are hung up. This ensures the hangup process + goes smoothly and no channels get hung in limbo. (issue #8088 + reported by kebl0155) ........ + +2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant + + * main/http.c: Add the Asterisk version information to the Server + header in HTTP responses. (requested by Pari) + + * include/asterisk/manager.h: Increase the maximum number of + manager headers to 128, at the request of Pari. + +2007-02-20 16:53 +0000 [r55555] Jason Parker + + * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free + with strdupa (thanks file) 55555! + +2007-02-20 16:41 +0000 [r55553] Russell Bryant + + * configs/sla.conf.sample: Change the formatting of sla.conf.sample + to make it more readable. (issue #9112, blitzrage) + +2007-02-19 21:12 +0000 [r55483] Olle Johansson + + * res/res_jabber.c: - Not sending arguments to an application is + not "out of memory" - Making error messages a bit more clear + +2007-02-19 18:11 +0000 [r55435] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) + | 2 lines forcename and forcegreetings options should check to + see if the recording already exists ........ + +2007-02-19 14:52 +0000 [r55397] Doug Bailey + + * channels/chan_iax2.c: Changed iax2 process thread to detached to + correct memory leak due to left over thread context on thread + exit. Modified module unload process to avoid deadlocks on + pthread cancels + +2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson + + * /, apps/app_record.c: Merged revisions 55277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 + lines Documentation update (#9053, jsmith) ........ + + * /: Block patch that was made only for 1.2 (already implemented in + 1.4 and trunk) + +2007-02-17 17:39 +0000 [r55219] Joshua Colp + + * apps/app_queue.c: Add missing membername option to AddQueueMember + documentation. (issue #9088 reported by seanbright) + +2007-02-17 17:10 +0000 [r55217] Jason Parker + + * channels/chan_skinny.c: Fix an issue where callerid would not be + displayed on some phones. Issue 8995, initial patch and research + done by wedhorn + +2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 + lines Answer the channel before recording privacy information. + (issue #8926 reported by lmamane) ........ + + * apps/app_queue.c: Make the 'i' option of Queue actually work. + (issue #8986 reported by utis) + + * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 + lines Allow chan_sip to handle attended transfers from a SIP + phone that is sitting behind chan_agent. Yes folks, all it took + was one line of code. (issue #8784 reported by pzieba) ........ + +2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in, configure.ac: If the + pg_config application is found, but there is probably executing + it, then consider postgres unavailable. (issue #8637) + + * codecs/gsm/Makefile: Filter out yet another architecture that + does not work with the optimizations in the built-in libgsm. + (issue 8637, ovi) + + * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged + revisions 55005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | + 9 lines Revert the change I did in revisions 54955, 54969, and + 54970, in 1.2, 1.4, and trunk. I decided that once a conference + is created from meetme.conf, it is acceptable behavior that the + pin can not be changed until the conference goes away. I also + added a note in meetme.conf to describe this behavior. We still + have another issue in 1.4 and trunk where some conferences with + no users don't go away. That is the real bug that needs to be + addressed here. ........ + +2007-02-16 22:18 +0000 [r55002] Joshua Colp + + * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb + 2007) | 2 lines Do not send indications through ast_indicate in + chan_agent but instead go directly to the technology. This way + when indications are emulated they happen on the Agent channel + and do not screw up formats on the channels. (issue #8439 + reported by punkgode) ........ + +2007-02-16 21:12 +0000 [r54969] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | + 5 lines For conferences that are configured in meetme.conf, check + the configuration file every time someone joins the conference + instead of only when the conference is first created. This is to + ensure that changes to the pin numbers in the config file are + always honored. (issue #9073) ........ + +2007-02-16 18:51 +0000 [r54924] Joshua Colp + + * apps/app_dial.c: Need to check macro extension as well as macro + context for directed pickup. + +2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant + + * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by + default. It was set to enabled in pbx.c. However, if the option + was not present in extensions.conf, then pbx_config.c would set + it back to disabled. + + * res/res_features.c: Clean up a few coding guidelines issues - + spaces to tabs, use sizeof() to pass the size of a static buffer, + add spaces ... + +2007-02-16 17:25 +0000 [r54886] Jason Parker + + * main/asterisk.c: Clarify a restart message. It's silly, but the + reporter had a very valid point. Issue 9079 + +2007-02-16 17:02 +0000 [r54884] Joshua Colp + + * apps/app_dial.c: Allow directed pickup to pick up the real + context instead of the macro context if a Macro is used. (issue + #8984 reported by jamesb63) + +2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson + + * channels/chan_sip.c: Issue #7541 - Handle multipart attachments + to SIP messages - even if boundary is quoted. + + * /, res/res_agi.c: Merged revisions 54771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 + lines Issue #9069 - If we open with TH we should not close with + /TD. (seanbright) ........ + +2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp + + * apps/app_speech_utils.c: Don't let dtmf leak over into the engine + and let it skew the results... also give DTMF results priority. + (issue #9014 reported by surftek) + + * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 + lines Use a separate variable to indicate execution should + continue instead of the return value. (issue #8842 reported by + pluto70) ........ + + * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue + #9068 reported by mhardeman) + +2007-02-14 18:44 +0000 [r54439] Olle Johansson + + * /: Block patch only needed in 1.2 + +2007-02-14 16:56 +0000 [r54375] Matt Frederickson + + * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 + lines When handling glare on a PRI, move the requested channel + rather than hang up the old one. Fix for 8957 and 9011. ........ + +2007-02-14 01:09 +0000 [r54290] Joshua Colp + + * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees + with it's placement, feel free to change it. (issue #9045 + reported by gork) + +2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant + + * channels/chan_sip.c: Remove a couple of leftover debug messages + + * include/asterisk/devicestate.h: Fix the documentation on the + return values from device state provider registration and + deletion. + + * channels/chan_sip.c: If we fail to create the SIP socket, then + return -1 from reload_config() so that load_module() will return + AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get + spammed with error messages every time chan_sip tries to send a + message. + +2007-02-13 18:41 +0000 [r54180] Olle Johansson + + * /: Blocking patch for 1.2 only + +2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant + + * main/dial.c, include/asterisk/dial.h: Change + ast_set_state_callback() to ast_dial_set_state_callback() + + * main/dial.c, apps/app_meetme.c, apps/app_page.c, + include/asterisk/dial.h: - Add the ability to register a callback + to monitor state changes in an asynchronous dial operation. - + Rename the various references to "status" to "state" in the dial + API + +2007-02-12 16:34 +0000 [r54026] Joshua Colp + + * configure, configure.ac: Make the --without-oss argument work. + (issue #9026 reported by puzzled) + +2007-02-12 15:38 +0000 [r54002] Russell Bryant + + * configs/users.conf.sample: Fix a typo where "vmpassword" should + be "vmsecret" + +2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach + + * channels/chan_h323.c: Fix VLDTMF reception + + * apps/app_echo.c: Much simpler than previous one ;-) + + * main/channel.c: Provide correct DTMF duration + + * main/cli.c: Bring deprecated 'debug channel ' command back + +2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming + + * configure, configure.ac, acinclude.m4: don't display the + --with-imap message unless --with-imap was specified without a + path use '-n' instead of '! -z' for tests + +2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant + + * apps/app_meetme.c: Add some output for "show application + SLAStation/SLATrunk" + + * channels/chan_sip.c: Change some text to properly state "On + Hold", which was already done in trunk. + + * configs/sla.conf.sample, include/asterisk/app.h, + include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, + channels/chan_sip.c, doc/sla.txt (added), + include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge + team/russell/sla_rewrite This is a completely new implementation + of the SLA functionality introduced in Asterisk 1.4. It is now + functional and ready for testing. However, I will be adding some + additional features over the next week, as well. For information + on how to set this up, see configs/sla.conf.sample and + doc/sla.txt. In addition to the changes in app_meetme.c for the + SLA implementation itself, this merge brings in various other + changes: chan_sip: - Add the ability to indicate HOLD state in + NOTIFY messages. - Queue HOLD and UNHOLD control frames even if + the channel is not bridged to another channel. linkedlists.h: - + Add support for rwlock based linked lists. dial.c: - Add the + ability to run ast_dial_start() without a reference channel to + inherit information from. + + * apps/app_echo.c: When the Echo() application receives the digit + '#', echo that back as well. Since we already sent the BEGIN + frame for that digit, it makes sense to send the END as well. + +2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming + + * channels/chan_gtalk.c: another dependency + + * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, + funcs/func_odbc.c, res/res_adsi.c: add some inter-module + dependencies + + * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk + scripts to work when both MODULEINFO and MAKEOPTS are present in + a source file + +2007-02-09 19:33 +0000 [r53749] Joshua Colp + + * apps/app_dial.c: Temporarily change musicclass on channel to one + specified in Dial so that the 'm' option functions properly. + (issue #8969 reported by christianbee) + +2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming + + * doc/imapstorage.txt, configure, configure.ac: clarify the fact + that voicemail IMAP storage cannot be built against a distro's + binary c-client library package (at least not at this time) + +2007-02-08 23:18 +0000 [r53672] Olle Johansson + + * main/acl.c: Don't output debug unless we asked for it + +2007-02-08 17:54 +0000 [r53601] Joshua Colp + + * apps/app_speech_utils.c: Fix timeout issue when utterance is + longer then timeout itself. + +2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher + + * main/loader.c: Issue 9007 - Mutex not released on early return + + * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) + | 2 lines Issue 9003 - If fullname is empty, quote() passes back + "\"" ........ + +2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant + + * main/db1-ast/Makefile: When building libdb1.a, put the additional + flags needed at the beginning of ASTCFLAGS, instead of at the + end. This way, we ensure that we find the local headers first + before accidentally trying to use headers that exist in locations + specified in the ASTCFLAGS passed from the main Makefile. (issue + #8637, ovi) + + * main/Makefile: The clean target actually needs to run "distclean" + on editline. This is because we need to make sure that its + configure script gets executed again, because the CFLAGS we want + to pass to editline may have changed. + +2007-02-07 17:53 +0000 [r53434] Joshua Colp + + * main/rtp.c: We can not reliably do P2P bridging with DTMF passing + back with compensation if we need to listen for DTMF frames. + (issue #8962 reported by caio1982) + +2007-02-07 17:39 +0000 [r53429] Russell Bryant + + * main/rtp.c: When parsing the NTP timestamp in a sender report + message, you are supposed to take the low 16 bits of the integer + part, and the high 16 bits of the fractional part. However, the + code here was erroneously taking the low 16 bits of the + fractional part. It then shifted the result 16 bits down, so the + result was always zero. This fix makes it grab the appropriate + high 16 bits, instead. (issue #8991, pointed out by + andre_abrantes) + +2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp + + * apps/app_playback.c: Directly load say.conf in load_module + instead of calling the reload function. (issue #8946 reported by + junky) + + * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 + lines Fix a few potential memory leaks with realtime users and + peers. (issue #8999 reported by bsmithurst) ........ + +2007-02-07 15:33 +0000 [r53355] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) + | 2 lines Issue 7440 - Macro called from Macro from the h + extension exits prematurely ........ + +2007-02-07 09:22 +0000 [r53324] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 52843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | + 1 line fixed some possible segfaults. also fixed an very + important bug which occurs on high load (when calls are very fast + generated) ........ + +2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher + + * res/res_jabber.c: Text fix for jabber reload command (reported by + bkruse via IRC) + + * main/manager.c, /: Merged revisions 53245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) + | 2 lines Issue 8987 - Status could return two responses + (mnicholson) ........ + +2007-02-05 23:43 +0000 [r53222] Olle Johansson + + * channels/chan_sip.c: Formatting + +2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp + + * apps/app_playback.c: Ensure say_cfg is NULL when the module is + loaded. (issue #8946 reported by junky) + + * apps/app_playback.c: Unregister Playback CLI commands as well as + dialplan application. (issue #8946 reported by junky) + +2007-02-05 00:18 +0000 [r53143] Olle Johansson + + * channels/chan_sip.c: Add some comments on queue system behaviour + and how it affects the SIP channel + +2007-02-03 21:05 +0000 [r53138] Joshua Colp + + * channels/chan_sip.c: Make SIPDtmfMode application work with + recent capability changes, and also fix an RTP stack issue when + the auto option was used. (issue #8972 reported by mdu113) + +2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant + + * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | + 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when + the dial application exits early because of invalid arguments + instead of just leaving it empty. (issue #8975) ........ + +2007-02-03 10:02 +0000 [r53131] Paul Cadach + + * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string + because due to compatibilities with CS1000 reported at + www.voip-info.org + +2007-02-02 21:26 +0000 [r53129] BJ Weschke + + * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a + warning to the console that things might possibly be + misconfigured when queue member's states are still 'Not in Use' + when we're about to bridge them with a caller from queue. Also, + put some documentation quoted from oej's queues.txt efforts + started in /trunk today. This commit puts #7433 into feedback + state for 1.4, and pending no further negative feedback, it will + finally be closed. + +2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp + + * main/rtp.c: Correct a copy/pasted error message line for RTCP. + + * main/config.c, /: Merged revisions 53117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 + lines Pass the glob expanded filename to process_text_line so + that error messages contain the actual filename, not the original + include one. (issue #8959 reported by tzafrir) ........ + + * Makefile: Add systemname to asterisk.conf generation per recent + discussions about it. (issue #8968 reported by blitzrage) + +2007-02-02 00:24 +0000 [r53109] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct + p2p RTP call setup in SIP. You can enable it in sip.conf, but it + is now considered experimental until we solve the + AST_CONTROL_ANSWER with payload and videocaps stuff. + +2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 + lines Copy noncodeccapability over to the joint variable so that + telephone-event will get transmitted in the sent INVITE. ........ + + * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile + here as well, but it apparently required both dev mode and no + optimizations to creep up. + + * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 + lines Don't negotiate RFC2833 when not configured to do so. + (issue #8799 reported by mdu113) ........ + +2007-02-01 21:24 +0000 [r53093] Russell Bryant + + * funcs/func_strings.c: Fix the FIELDQTY function to not crash. + (reported by blitzrage and Corydon on IRC) + +2007-02-01 21:15 +0000 [r53091] Olle Johansson + + * /: Going backwards, blame file. + +2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp + + * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb + 2007) | 2 lines Return previous behavior of having MOH pick up + where it was left off. (issue #8672 reported by sinistermidget) + ........ + + * funcs/func_strings.c: Make func_strings build under dev mode. + Didn't I do this today already in the berkeley DB? + +2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson + + * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement + call counter - If it's still set at time of dialog destruction, + make sure we decrement the device call counter properly before we + destroy the dialog + + * apps/app_queue.c: Change debug level for state change message + that is not really informative when debugging app_queue + + * channels/chan_sip.c: Cleaning up the devicestate callback + function + +2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher + + * funcs/func_strings.c: Oops. + + * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) + | 2 lines Bug 8965 ........ + +2007-02-01 19:33 +0000 [r53072] Joshua Colp + + * main/asterisk.c: Add missing 'F' letter to getopt so it magically + becomes a valid option. (issue #8960 reported by tzafrir) + +2007-02-01 19:21 +0000 [r53070] Tilghman Lesher + + * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) + | 2 lines No wonder FIELDQTY doesn't work with functions... the + documentation in pbx.c was wrong ........ + +2007-02-01 17:37 +0000 [r53064] Joshua Colp + + * channels/chan_sip.c: Fix silly logic. We really want to write + UDPTL frames out when the call is up. + +2007-02-01 16:35 +0000 [r53062] Olle Johansson + + * configs/sip.conf.sample: Add explanation of port= in combination + with defaultip= (thanks jsmith) + +2007-02-01 13:17 +0000 [r53060] Christian Richter + + * channels/chan_misdn.c: we update the name on any first reply of + our setup + +2007-02-01 11:07 +0000 [r53057] Paul Cadach + + * channels/chan_h323.c: chan_h323 is very stable, so let it built + by default + +2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp + + * main/rtp.c: When going on hold have the side that was put on hold + reinvite back to Asterisk. When going off hold have the side that + was taken off hold reinvited back to the other party. + + * main/rtp.c: Add more frame types to forward in the RTP bridge + loops. + +2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant + + * main/cdr.c, main/manager.c, pbx/pbx_spool.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, + channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: + Merged revisions 53045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | + 3 lines Fix a bunch of places where pthread_attr_init() was + called, but pthread_attr_destroy() was not. ........ + + * apps/app_userevent.c: Remove an extra \r\n from manager user + events. (issue #8955, mnicholson) + + * main/rtp.c, /: Merged revisions 53039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | + 3 lines Use the proper format string to print unsigned values in + the rtp debug output. (issue #8954, wmis) ........ + + * apps/app_queue.c: Only changed the paused status in an existing + queue member if the paused column exists. + + * apps/app_queue.c: Instead of always creating a realtime queue + member as unpaused, read the "paused" column and use that value + for the paused status of the member. (issue #8949, jmls) + + * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. + (issue #8363, johnlange) + + * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue + #8942, lters) + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + codecs/codec_gsm.c: When we are checking for a system installed + version of libgsm, we need to check for gsm.h as well. + Furthermore, when checking for this header, it may be located in + a gsm/ sub directory, so check for that, as well. (issue #8773) + + * channels/chan_sip.c: Only set the DTMF flag on the rtp structure + if the DTMF mode is actually RFC2833, not just that it is not + INFO. This makes it get set for inband DTMF as well, which is not + valid. (issue #8936) + + * main/asterisk.c, /: Merged revisions 52903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | + 9 lines The SIGHUP handler was implemented to allow admins to + send SIGHUP to a running Asterisk process to reload the + configuration. However, doing the actual reload in the signal + handler itself is a very bad thing to do, because the reload + process includes calling non-reentrant functions such as + malloc/calloc/etc. If Asterisk is running in the background, then + the reload will happen immediately. However, if running in + console mode, the reload doesn't work until something is typed at + the console. That sort of defeats the purpose, but I don't see an + easy way to get around it at this point. ........ + +2007-01-30 15:29 +0000 [r52856] Joshua Colp + + * channels/chan_iax2.c: Drop the deprecated show commands since the + original ones were changed back. (issue #8937 reported by + PCadach) + +2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach + + * channels/chan_h323.c: Revert reprecation of h.323 gk cycle + command from pre-1.4 version instead of duplicated h323 cycle gk + + * res/res_odbc.c: Don't play with free()'d pointers + + * configure, acinclude.m4: Handle non-standard OpenH323/PWLib + library names + +2007-01-30 00:15 +0000 [r52763] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | + 5 lines Fix the extraction of the timestamp from video frames. It + was using the mapping for a mini-frame instead of a video-frame, + which caused it to get invalid data. (issue #8795, mihai) + ........ + +2007-01-29 23:43 +0000 [r52717] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan + 2007) | 2 lines Now that filename is part of the structure and + since it comes before postprocess... we have to add it to our + postprocess line. (reported on asterisk-dev by Boris Bakchiev) + ........ + +2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant + + * main/Makefile: Add a missing quotation mark. This was pointed out + by jcmoore on #asterisk-dev. + + * main/manager.c: Remove a recursive lock of the manager session. + This was pointed out by zandbelt in issue #8711. + +2007-01-29 22:12 +0000 [r52679] Tilghman Lesher + + * pbx/pbx_config.c: Argument number correction + +2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant + + * main/Makefile: ASTLDFLAGS needs to be passed to the editline + configure script as LDFLAGS. (issue #8928, zandbelt) + + * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF + mode translation. P2P bridging can only be used when the DTMF + modes don't match if the core is monitoring DTMF in both + directions. Then, the core will handle the translation. + Otherwise, this bridging method can not be used. (issue #8936) + + * main/manager.c: The session lock can not be held while calling + action callbacks. If so, then when the WaitEvent callback gets + called, then no event can happen because the session can't be + locked by another thread. Also, the session needs to be locked in + the HTTP callback when it reads out the output string. This fixes + the deadlock reported in both 8711 and 8934. Regarding issue + 8711, there still may be an issue. If there is a second action + requested before the processing of the first action is finished, + there could still be some corruption of the output string buffer + used to build the result. (issue #8711, #8934) + +2007-01-29 18:59 +0000 [r52572] Joshua Colp + + * apps/app_voicemail.c: Use ast_calloc instead of malloc. + +2007-01-29 17:57 +0000 [r52535] Steve Murphy + + * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR + backport to 1.4). It was committed to trunk via 7663. But it + wasn't so much an enhancement as a fix for the bad language + output for portuguese in Brazil, so, after a lot of prodding from + patient Brazilians, here is the same fix for 1.4 + +2007-01-29 17:33 +0000 [r52523] Joshua Colp + + * apps/app_voicemail.c: Set quota information to 0 when creating a + vm_state. (issue #8924 reported by neutrino88) + +2007-01-29 16:54 +0000 [r52506] Russell Bryant + + * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in + the last commit to the adaptive jitterbuffer code. - Specifically + indicate to the compiler that the "dropem" variable only needs + one but. - Change formatting to conform to coding guidelines. + +2007-01-29 04:18 +0000 [r52494] Jim Dixon + + * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with + jitterbuf, whereas it would not complain about, and would allow + itself to be overfilled (per the max_jitterbuf parameter). Now it + rejects any data over and above that size, and complains about + it. + +2007-01-28 05:15 +0000 [r52462] Tilghman Lesher + + * configure, configure.ac: Suggested change to fix normal usage of + --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing + list) + +2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp + + * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 + lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log + follow documentation. (issue #7677 reported by amilcar) ........ + + * main/manager.c: Have the manager interface send back an "Already + logged in" message instead of "Invalid/Unknown Command" when the + client authenticates for a second time. (issue #8509 reported by + pari) + + * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 + lines Make the last context entry read in the dominant one. + (issue #8918 reported by pj) ........ + + * main/file.c: Fix core show file formats CLI command. + +2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp + + * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 + lines Allow dequeueing of frames with negative timestamp by + moving jitterbuffer frames check to jb_next. (issue #8546 + reported by harmen) ........ + + * channels/chan_sip.c: Drop out variables I accidentally put in. + + * channels/chan_sip.c: Decrement onHold count if we are hung up on + and still on hold. (issue #8909 reported by alexh42) + + * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan + 2007) | 2 lines Add another note about audio files being played + back to each bridged party. (issue #8718 reported by ppyy) + ........ + +2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant + + * apps/app_voicemail.c, configs/users.conf.sample: By suggestion + from kpfleming last week, change "vmpassword" to "vmsecret". + + * configure, configure.ac: Remove libnsl as a required lib for + libiksemel to work. This change was already made in the trunk. + (issue #8762) + + * include/asterisk/dial.h: Fix the formatting of doxygen comments + to properly indicate that the comment documents the previous + entity, as opposed to the next one. + +2007-01-24 18:26 +0000 [r52052] Steve Murphy + + * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 + line updated check_expr via 8322 (refactoring of expression + checking impl); elfring contributed a nice code reorg, I + contributed some time to get it working again, better messages + ........ + +2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp + + * main/dial.c (added), apps/app_page.c, main/Makefile, + include/asterisk/dial.h (added): Merge in dialing API and the + app_page that uses it. (issue #BE-118) + + * channels/chan_sip.c: Fix changing channel formats when joint + capability changes and there are no audio formats... I didn't + break it originally! (issue #8535 reported by ivoc) + +2007-01-24 17:14 +0000 [r52000] Russell Bryant + + * configure: rebuild configure script to reflect last chan_h323 + related changes. + +2007-01-24 12:57 +0000 [r51979-51989] Christian Richter + + * channels/chan_misdn.c: added fix from #8899 + + * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 + Jan 2007) | 1 line fixed the busy problem (dialstatus was not + busy when we called a busy extension) ........ + +2007-01-24 09:30 +0000 [r51931] Olle Johansson + + * channels/chan_sip.c: Show capabilities *and* preference in + general settings in "sip show settings" (reported by Clona/Telio + - Thanks!) + +2007-01-24 08:04 +0000 [r51895] Paul Cadach + + * acinclude.m4: Allow x64 builds of H.323 (please, rebuild + configure) + +2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant + + * main/channel.c, /: Merged revisions 51843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | + 6 lines Fix an issue related to synchronization of recordings + when using Monitor(). The bug is a miscalculation of the amount + to seek the stream for writing to disk when the number of samples + coming in and out of a channel do not match up. (issue #8298, + #8887, report and patch by guillecabeza, patch files created and + testing done by whoiswes) ........ + + * apps/app_while.c, /: Merged revisions 51828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | + 4 lines Don't set a new value for the END_ variable on the + channel before using the old value. If you do, it will lead to + accessing a memory address that has been free()'d. (issue #8895, + arkadia) ........ + +2007-01-23 22:46 +0000 [r51788] Joshua Colp + + * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, + channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_alsa.c, + channels/chan_gtalk.c, channels/chan_iax2.c: Update channel + drivers to use module referencing so that unloading them while in + use will not result in crashes. (issue #8897 reported by junky) + +2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant + + * main/manager.c: Fix some bugs in process_message(). The manager + session lock needs to be held when sending some sort of response, + or calling one of the manager action callbacks. This resolves an + issue where people using the GUI would get random crashes when + they start clicking around a lot. (issue #8711, reported and + debugged by zandbelt) + + * main/http.c: Fix setting the default port of 8088 on 64-bit or + big-endian machines. + + * main/manager.c: When traversing the list of manager actions, the + iterator needs to be initialized to the list head *after* locking + the list. Also, lock the actions list in one place it is being + accessed where it was not being done. + +2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy + + * res/res_features.c: this mod from 8593 (dstchannel in cdr is + empty when transfer call). + + * main/callerid.c: via 8748 (callerid.c loses name when returning + PRIVATE_NUMBER flag), the user suggested this mod, saying it + would allow 'WITHHELD' to appear in the name field, which would + be useful + +2007-01-23 10:28 +0000 [r51648-51649] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | + 6 lines * more additions to make the RESTART message work * added + fix for misdn_call to allow SETUPs with empty extensions, + replaced the strtok_r functions with strsep for that (inspired by + Sandro Cappellazzo, thanks) ........ r50506 | crichter | + 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get + L2 UP, the L1 is UP definitely too, so we set the L1 state up as + well. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: manually merged r49922 and r50335, because + of conflicts. this commint includes addition of the ISDN RESTART + Message + +2007-01-23 06:51 +0000 [r51615] Paul Cadach + + * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk + startup if h323 configuration file not found (reported by + mithraen) + +2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp + + * channels/chan_sip.c: Only change audio formats on the channel if + we have an audio format to change to. (issue #8535 reported by + ivoc) + + * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan + 2007) | 2 lines Yield before reading from zaptel timing source + under Solaris so that other threads get a chance to do things. + (issue #7875 reported by bob) ........ + +2007-01-22 19:28 +0000 [r51409] Steve Murphy + + * pbx/pbx_ael.c: This fixes 8836, according to dnatural + +2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan + 2007) | 2 lines Move filestream creation to Mixmonitor loop. This + will prevent a blank file from being created if no frames ever + pass through to be recorded. (issue #7589 reported by + steve_mcneil) ........ + +2007-01-20 06:53 +0000 [r51348-51350] Jason Parker + + * configs/say.conf.sample: Fix Italian numeral support in say.conf + for "_[2-9]00" case. "2131" would've translated to something + along the lines of (pardon my..Italian {or lack thereof}) + "duecentocentotrentuno", which makes no sense at all. + + * configs/say.conf.sample: Fix German language support in say.conf + Properly support 21, 31, 41, 51, 61, 71, 81, and 91. + einundzwanzig has the same format as zweiundzwanzig (as do all + other "_ZX" spoken numerals) Fix support for numbers in the + 10,000,000 to 99,999,999 range. Add support for numbers in the + 100,000,000 to 999,999,999 range. + +2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant + + * apps/app_meetme.c: Remove an unused instance of an unnamed enum. + + * apps/app_meetme.c: Remove another duplicated definition + + * apps/app_meetme.c: Remove a variable that was declared twice. + + * codecs/gsm/Makefile: Add a couple more processors that need + optimizations excluded. (issue #8637) + + * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. + AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same + thing. So, a digit would have been interpreted incorrectly here. + Since the channel driver will always have the begin and end + callbacks called for a digit, only support the button-down and + button-up messages. + + * .cleancount: Bump the cleancount since my last commit changed the + channel structure. + + * channels/chan_oss.c, main/rtp.c, main/channel.c, + channels/chan_phone.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_features.c, + channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, + channels/chan_zap.c, channels/chan_local.c, main/frame.c, + channels/chan_sip.c, channels/chan_agent.c, + include/asterisk/channel.h, channels/chan_gtalk.c, + channels/chan_iax2.c: Merge the changes from the + /team/group/vldtmf_fixup branch. The main bug being addressed + here is a problem introduced when two SIP channels using SIP INFO + dtmf have their media directly bridged. So, when a DTMF END frame + comes into Asterisk from an incoming INFO message, Asterisk would + try to emulate a digit of some length by first sending a DTMF + BEGIN frame and sending a DTMF END later timed off of incoming + audio. However, since there was no audio coming in, the DTMF_END + was never generated. This caused DTMF based features to no longer + work. To fix this, the core now knows when a channel doesn't care + about DTMF BEGIN frames (such as a SIP channel sending INFO + dtmf). If this is the case, then Asterisk will not emulate a + digit of some length, and will instead just pass through the + single DTMF END event. Channel drivers also now get passed the + length of the digit to their digit_end callback. This improves + SIP INFO support even further by enabling us to put the real + digit duration in the INFO message instead of a hard coded 250ms. + Also, for an incoming INFO message, the duration is read from the + frame and passed into the core instead of just getting ignored. + (issue #8597, maybe others...) + + * main/asterisk.c: Merged revisions 51300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | + 4 lines Fix a memory leak on command line tab completion. The + container for the matches was freed, but the individual matches + themselves were not. (issue #8851, arkadia) ........ + +2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard + + * channels/chan_zap.c: chan_zap compiles without libpri after + committing 7877 patch + + * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) + | 3 lines issue 7877: chan_zap module reload does not use + default/initialized values on subsequent loads. Reset + configuration variables to default values prior to parsing + configuration file. ........ + +2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming + + * /: block this patch since it is already here + +2007-01-18 22:50 +0000 [r51265] Jason Parker + + * apps/app_voicemail.c, main/channel.c, main/pbx.c, + funcs/func_strings.c, main/app.c: Add some more checks for + option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, + patch(es) by tgrman + +2007-01-18 21:54 +0000 [r51262] Russell Bryant + + * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: + Ensure that the locations given to the Asterisk configure script + for ncurses, curses, termcap, or tinfo are further passed along + to the editline configure script. This fixes some + cross-compilation environments. (issue #8637, reported by ovi, + patch by me) + +2007-01-18 21:14 +0000 [r51256] Tilghman Lesher + + * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 + Jan 2007) | 2 lines If a timezone is not specified, assume + localtime (instead of gmtime) (Issue #7748) ........ + +2007-01-18 19:17 +0000 [r51251] Joshua Colp + + * apps/app_speech_utils.c: Only start timeout once we reach the end + of the files to play back. + +2007-01-18 18:42 +0000 [r51245] Jason Parker + + * main/cli.c: Fix an issue with file name completion in "module + load" and "load". Issue 8846 + +2007-01-18 18:36 +0000 [r51243] Joshua Colp + + * channels/chan_sip.c: Copy MOH settings when calling a peer so + that if they put someone on hold or get put on hold themselves + they get the right music class. (issue #8840 reported by mdu113) + +2007-01-18 18:28 +0000 [r51241] Jason Parker + + * main/channel.c: Fix an issue with deprecated commands + +2007-01-18 17:49 +0000 [r51236] Tilghman Lesher + + * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 + Jan 2007) | 2 lines Document all the fields, including the + indication that "uniqueid" should not be renamed. ........ + +2007-01-18 17:18 +0000 [r51233] Russell Bryant + + * main/manager.c: Make the "hasmanager" option in users.conf + actually have an effect. (issue #8740, LnxPrgr3) + +2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp + + * apps/app_voicemail.c: Build the IMAP remote directory string + better and properly. Fix an issue with encoding the GSM voicemail + when attaching to the voicemail. (issue #8808 reported by + akohlsmith) + + * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. + (issue #8840 reported by mdu113) + +2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant + + * funcs/func_odbc.c: Fix some instances where when loading + func_odbc, a double-free could occur. Also, remove an unneeded + error message. If the failure condition is actually a memory + allocation failure, a log message will already be generated + automatically. + + * channels/chan_zap.c: Instead of dividing the offset by 2 + directly, make it more clear that the offset is being scaled by + the size of the elements in the buffer. (Inspired by a discussing + on the asterisk-dev list about this code) + + * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | + 3 lines Move the check for a failure of ast_channel_alloc() to + before locking the pvt structure again. Otherwise, on a failure, + this will cause a deadlock. ........ + +2007-01-17 20:56 +0000 [r51195] Tilghman Lesher + + * /, main/utils.c: Merged revisions 51194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) + | 4 lines When ast_strip_quoted was called with a zero-length + string, it would treat a NULL as if it were the quoting character + (and would thus return the string in memory immediately following + the passed-in string). ........ + +2007-01-17 17:36 +0000 [r51186] Jason Parker + + * apps/app_voicemail.c: re-add "password" for realtime voicemail + +2007-01-17 06:36 +0000 [r51182] Joshua Colp + + * main/rtp.c: Return the correct result when directly writing out a + packet so that the core doesn't then decide to handle it the + regular way again. (issue #8833 reported by rcourtna) + +2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming + + * apps/app_voicemail.c: a few more coding style cleanups and one + bug fix (from AnthonyL) + +2007-01-17 00:46 +0000 [r51172] Joshua Colp + + * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the + scheduler callback. + +2007-01-17 00:20 +0000 [r51165-51170] Jason Parker + + * main/rtp.c: Fix issue with dtmf continuation packets when the + dtmf digit is 0... Issue 8831 + + * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with + IMAP storage and realtime voicemail. Also update the vmdb sql + script for IMAP specific options. Issue 8819, initial patches by + bsmithurst (slightly modified by me) + + * doc/voicemail_odbc_postgresql.txt: change documentation to + reflect new procedure in 1.4/trunk + +2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher + + * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions + 51161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) + | 2 lines Add documentation walkthrough on getting Postgres to + work with voicemail (from Issue 8513) ........ + + * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) + | 2 lines Postgres driver doesn't like a NULL pointer when + retrieving the length (Bug 8513) ........ + +2007-01-16 17:46 +0000 [r51150] Matt O'Gorman + + * apps/app_voicemail.c: minor things i missed before i get jumped + on + +2007-01-16 17:39 +0000 [r51148] Joshua Colp + + * /, res/res_features.c: Merged revisions 51145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 + lines Return previous behavior. ParkedCalls will be able to do + DTMF based transfers again. trunk however will get an option to + allow this to be set on/off. (issue #8804 reported by nortex) + ........ + +2007-01-16 17:36 +0000 [r51146] Jason Parker + + * main/file.c: Display more useful output when streaming files. + Include the channel name to which the file is being played. Issue + 8828, patch by junky. + +2007-01-16 05:55 +0000 [r51087] Joshua Colp + + * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 + lines Add none as a valid callgroup/pickupgroup option. I + consider it a bug that it would inherit it all the way down and + not have any way to reset it to nothing - so that's why it is in + 1.2. (issue #8296 reported by gkloepfer) ........ + +2007-01-16 01:15 +0000 [r51057] Russell Bryant + + * main/config.c: It is possible for the config pointer to be NULL + here, so it needs to be checked before dereferencing it. + +2007-01-16 00:22 +0000 [r51030] Matt O'Gorman + + * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for + changing voicemail password in users.conf from voicemail main, + written by AnthonyL bug #8436 + +2007-01-15 23:49 +0000 [r50994] Russell Bryant + + * Makefile.rules: Filter out a few CFLAGS that are not valid + CXXFLAGS. + +2007-01-15 21:08 +0000 [r50957] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 + | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 + lines Solves issue with forwarding voicemails from folders other + than inbox. patch by anthonyl. ........ + +2007-01-15 18:23 +0000 [r50921] Jason Parker + + * main/asterisk.c: re-add deprecated "show version" CLI command. + +2007-01-15 16:36 +0000 [r50895] Joshua Colp + + * main/manager.c: Move event processing into do_message so that it + gets executed again when events are tripped. + +2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, main/Makefile, + configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the + ACX_PTHREAD macro from the Autoconf macro archive for setting up + compiler pthreads support... should improve portability to + platforms with unusual pthreads requirements + +2007-01-14 21:59 +0000 [r50820] Joshua Colp + + * main/astmm.c: Add missing newlines for two memory CLI commands. + +2007-01-14 05:13 +0000 [r50782] Tilghman Lesher + + * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, + main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, + main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, + main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, + main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, + main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, + main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, + main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, + main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, + main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, + main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, + main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, + main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, + main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, + main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, + main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 + Jan 2007) | 2 lines Bug 8814 - db should look for its header + using a relative path, instead of the system path (Fixes FreeWRT) + ........ + +2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming + + * Makefile, build_tools/make_sample_voicemail (added): when + building the sample greetings for maibox 1234@default during + 'make samples', build a greeting for each language and file + format the user selected to install with menuselect (reported by + Brian Capouch on asterisk-dev) + +2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp + + * main/channel.c: Only write a frame out to the channel if one + exists. There are cases where one may not and would therefore + cause the channel driver to segfault. (issue #8434 reported by + slimey) + + * res/res_snmp.c: Only join the snmp thread on an unload if the + thread is actually running. (issue #8810 reported by junky) + +2007-01-12 19:24 +0000 [r50647] Jason Parker + + * configs/voicemail.conf.sample: Update documentation to state that + you shouldn't use realtime static with voicemail.conf + +2007-01-12 16:42 +0000 [r50602] Joshua Colp + + * main/manager.c: We need to check for res being 0 in do_message + itself, otherwise our headers will get lost. + +2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming + + * main/pbx.c, /: Merged revisions 50561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) + | 2 lines minor documentation clarification ........ + +2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp + + * channels/chan_sip.c: Remove check for channel state as it can + definitely be something other then ring, and also clean up the + code a bit. This should solve the parking issues and maybe some + attended transfer issues people have been seeing. + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add + support to see whether NAT was detected (yay symmetric RTP) and + also add a check in chan_sip so that if NAT has been detected and + the reinvite behind nat option has been turned off, then just do + partial bridge. (issue #8655 reported by mnicholson) + + * apps/app_speech_utils.c: Merge speech-multi branch which adds + support for joining multiple sound files together to be played + one after another in SpeechBackground. + + * main/config.c: Fix parsing when using something like ldap + settings. (done by anthonyl) + + * channels/chan_sip.c: Fix chan_sip not working issue. Let's not + prematurely return 0. (issue #8783 reported by st41ker) + +2007-01-10 16:45 +0000 [r50346] Jason Parker + + * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made + it fail to load if the config file existed. Issue 8777 + +2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 + lines Add another return value to dial_exec_full that indicates + execution is going to continuing at a new + extension/context/priority and to just let it slide. (issue #8598 + reported by jon) ........ + + * main/pbx.c: Ensure data's existence before trying to access it. + (issue #8774 reported by rcourtna) + +2007-01-10 02:17 +0000 [r50228] Russell Bryant + + * Makefile, /: Merged revisions 50227 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | + 6 lines Make the number that represents the major version number + a single digit instead of 2. Using two digits makes it an octal + number when put into version.h, which breaks the compilation of + any out of tree module that checks the version for any version + after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev + mailing list, who gave credit to vihai for pointing it out) + ........ + +2007-01-09 17:11 +0000 [r50186] Jason Parker + + * main/cli.c: Re-add CLI command that should have only been + deprecated in 1.4. Thanks kshumard! (reported in person, so no + associated issue #) + +2007-01-09 13:40 +0000 [r50151] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) + | 4 lines The advent of realtime has enabled people to use commas + in the fullname field. This could cause an issue with sending + voicemails, when the field is unquoted. (Issue 8595) ........ + +2007-01-09 11:25 +0000 [r50124] Olle Johansson + + * channels/chan_sip.c: - handle re-invites properly in sip_hangup() + - Add some invitestate status changes just to be sure + +2007-01-08 23:39 +0000 [r50098] Jason Parker + + * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, + where it wouldn't ever parse a password, since it was using + "secret" instead of "password" Issue 8761, reported by and patch + suggestion from ssokol. + +2007-01-08 21:11 +0000 [r50073] Matt O'Gorman + + * apps/app_senddtmf.c: we can't unlock a channel if we cant find + it. - AnthonyL bug #8741 + +2007-01-08 18:21 +0000 [r50032] Joshua Colp + + * main/rtp.c: Disable the more intense packet2packet bridging until + the bugs can be worked out. + +2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson + + * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 + re-invite This is not a fix, but adding an error message to tell + the admin that we have a bad configuration. We should not send + T.38 re-invites to devices that can't handle it (with the current + architecture where you have to hard-code t.38 support per + device). To really fix this, we need to figure out a way to tell + the incoming call that the re-invite failed, so we can signal + failure on that end and go back to the original call. + + * channels/chan_sip.c: Issue #8524, support multiple via header + values (tardieu) Thanks! + + * channels/chan_sip.c: We only need one forward declaration + + * channels/chan_sip.c: Issue 8735: Terminate state when extension + is unavailable for subscription + +2007-01-08 05:11 +0000 [r49890] Joshua Colp + + * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 + lines Ensure we use the default refresh value of 60 if the remote + server does not send one. (issue #8746 reported by maethor) + ........ + +2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming + + * configure, configure.ac: since we use AC_PATH_TOOL to find tools, + we should use the results it provides for us (reported by Brian + Capouch on the asterisk-dev list) + +2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher + + * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) + | 2 lines If openstream fails, then we crash (Issue 8564) + ........ + + * channels/chan_sip.c: Second condition was a subset of the first, + so hold was never decremented, thus hint stayed stuck (Issue + 8747) + +2007-01-06 00:24 +0000 [r49742] Jason Parker + + * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping + byte of allocated memory! This looks like it may have been a + chicken/egg scenario.. You had to call a cleanup func, because + everything was allocated. Then since you had to call a cleanup + func, you were forced to allocate - ie; strdup(""). + +2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming + + * configure, acinclude.m4: one more time... + + * configure, acinclude.m4: proper fix for r49712 + + * configure, acinclude.m4: if --with-foo= is specific for a + configure option, ensure that it is used for header file checking + as well + + * main/manager.c: ast_func_read() needs a writable copy of the + function name to be passed + +2007-01-05 23:16 +0000 [r49705] Jason Parker + + * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and + chan_zap also depend on zaptel. This fixes an issue (8727) with + zaptel being in a different directory, using --with-zaptel. + +2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming + + * main/manager.c: don't 'consume' the params list before we try to + use it again + + * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, + main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, + main/db.c, channels/chan_zap.c, channels/chan_sip.c, + apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, + utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, + apps/app_queue.c, res/res_jabber.c: reduce stack consumption for + AMI and AMI/HTTP requests by nearly 20K in most cases + +2007-01-05 22:14 +0000 [r49675] Joshua Colp + + * main/channel.c: Don't keep repeating the warning over and over + when the end of the call is reached. (issue #8724 reported by + xrg) + +2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming + + * /, channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_iax2.c: Merged revisions 49635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) + | 2 lines ensure that threads which are supposed to be detached + (because we aren't going to wait on them) are created properly + ........ + + * channels/chan_iax2.c: revert the dynamic_list insertion change... + that was not the right thing to do + + * channels/chan_iax2.c: create the IAX2 processing threads as + background threads so they will use smaller stacks when we create + a dynamic thread, put it on the dynamic_list right away so we + don't lose track of it + +2007-01-04 23:00 +0000 [r49568] Joshua Colp + + * channels/chan_iax2.c: It's possible for the iax2 pvt to + disappear, so if it has... don't bother looking for dpentries. + +2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming + + * include/asterisk/threadstorage.h, main/asterisk.c, + build_tools/cflags.xml, include/asterisk.h, main/Makefile, + main/threadstorage.c (added), main/utils.c: add support for + tracking thread-local-storage objects that exist via + 'threadstorage' CLI commands + +2007-01-04 22:28 +0000 [r49551] Joshua Colp + + * main/config.c: Only free comments and line buffer once we reach + the first level. (issue #8678 reported by ssokol, fixed by + anthonyl) + +2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming + + * channels/iax2-parser.c, main/frame.c: don't mark these + allocations as 'cache' allocations when caching has been disabled + + * channels/iax2-parser.c: if we're going to decrement the frame + count when we free a frame, we should inrement it when we create + one :-) + + * channels/iax2-parser.c, channels/iax2-parser.h, + channels/chan_iax2.c: only do IAX2 frame caching for voice and + video frames + + * main/frame.c: don't do frame header caching in the core if + LOW_MEMORY is defined + + * channels/iax2-parser.c: don't define this type either if + LOW_MEMORY is enabled + +2007-01-04 18:11 +0000 [r49459] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 + | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 + lines converted a lot of 256 to PATH_MAX and some white space + fixes. ........ + +2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming + + * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode + + * codecs/Makefile: make building of codec_gsm against the system + GSM library actually work + +2007-01-04 16:50 +0000 [r49413] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 + | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 + lines good catch russell sorry i missed that. fix magic number + with proper sizeof ........ + +2007-01-04 04:33 +0000 [r49388] Russell Bryant + + * funcs/func_realtime.c: Fix the REALTIME() dialplan function. + ast_build_string() advances the string pointer to the position to + begin the next write into the buffer. So, this pointer can not be + used to copy the contents of the string later. The beginning of + the buffer must be saved. Interestingly enough, this code could + not have ever worked. (Pointed out by Sebb on IRC, thanks!) + +2007-01-03 23:32 +0000 [r49355] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 + | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 + lines When using ODBC_STORAGE VoicemailMain doesn't create the + subdirectories for a mailbox such as the INBOX directory. this + patch solves that problem, was written by anthony be-125 ........ + +2007-01-03 09:06 +0000 [r49313] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, + /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, + configs/misdn.conf.sample: Merged revisions + 48319,48321,48467,48552,48576,49135,49303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | + 1 line changed a few debugs to higher debug levels ........ + r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | + 1 line added the export and import of the MISDN_ADDRESS_COMPLETE + Variable to inidcate wether the extension is already completely + dialed or if there might come additional digits by information + elements. also added some docs for that. ........ r48467 | + crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line + removed FIXUP state. added check for channel allocation conflict + when we create a setup while the other site creates a setup on + the same channel, besides the check we resolve this conflict. + ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 + Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a + preselected channel we just accept it, even when we're NT. added + some checks for segfaults. ........ r48576 | crichter | + 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we + reject a channel, because it's in use already, we shouldn't + process the setup anymore. made the channel allocation a bit + easier and more understandable, removed a few unused lines + ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 + Jan 2007) | 1 line added check for channel ranges in the + set/empty channel functions. set pmp_l1_check default to no. + added misdn restart pid cli command. added cleaning of channel + when we send a RELEASE_COMPLETE. ........ r49303 | crichter | + 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added + check for bridging in misdn_call to avoid setting + echocancellation when 2 mISDN channels are involved and when + bridging is set. That lead to a kernel panic before under + different situations, because we switched about 2 times between + hardware bridging and echocancelation * readded MISDN_URATE + variable which got lost before, this should make app_v110 work + again * fixed typo ........ + +2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming + + * Makefile, Makefile.rules: various Makefile improvements to get + chan_vpb (and any other C++ modules) to build properly + +2007-01-03 01:19 +0000 [r49259] Joshua Colp + + * channels/chan_iax2.c: Check pvt structure presence before passing + to send_command. This gets rid of the irritating message about a + packet without pvt structure. This happens because the scheduled + item is getting cancelled at almost the exact moment it is + getting executed. + +2007-01-02 22:30 +0000 [r49237] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, + pbx/ael/ael.flex: This is a slight modification to Josh's edits + for #8579; both files edited were the produced by flex; so the + source files need to be changed instead, and the generated files + regenerated. + +2007-01-02 19:58 +0000 [r49212] Olle Johansson + + * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always + enabled at that point in the code + +2007-01-02 17:33 +0000 [r49189] Jason Parker + + * main/pbx.c: Allow fractions of a second in the Wait() + application, like it says it allows. + +2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming + + * channels/chan_zap.c: remove comment that is unrelated to this + function + +2007-01-02 12:08 +0000 [r49145] Olle Johansson + + * configs/features.conf.sample: Adding note on effect of + applicationmap features on re-invites + +2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming + + * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, + configure.ac, codecs/codec_zap.c: check specifically for VLDTMF + and transcoding support in the system's Zaptel installation, and + make only the modules that need those features dependent on them + (this will allow building the other Zaptel-using parts of + Asterisk against older versions of Zaptel or those on other + platforms that haven't caught up yet to the Linux version) + + * Makefile: use a simpler (and portable) method to ensure that + menuselect is built as a host binary + + * Makefile: revert this change until a better solution can be + found... 'env -i' was not being used properly, but even when + changed to do so, this process fails during cross-compilation + because the menuselect build still sees 'CC' as set to the + cross-compiler + +2007-01-01 20:14 +0000 [r49096] Olle Johansson + + * channels/chan_sip.c: remove incomplete implementation of dnsmgr. + Let's fix this in trunk. + +2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp + + * pbx/pbx_config.c: IAX has been deprecated for quite some time so + we had better use IAX2 when creating the dial string for users. + (issue #8697 reported by ssokol) + + * channels/chan_zap.c: Use asprintf to build the channel names + instead of custom function. I believe the custom function is + doing some things that are not portable across all + implementations. (issue #8570 reported by hterag & issue #8692 + reported by nicolasg) + + * main/rtp.c: If the Packet2Packet bridge is being broken because + of a masquerade then attempt to read a frame in so the masquerade + actually happens. Otherwise weirdness will occur. (issue #8696 + reported by kjotte) + + * channels/chan_iax2.c: Initialize the packet queue in load_module + instead of just declaring the list with the default value. (issue + #8695 reported by ssokol) + +2006-12-30 00:40 +0000 [r49061] Steve Murphy + + * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have + comma args converted to vertical bars. I hope this change does + little harm. + +2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming + + * /: put this value into the correct property + + * /, BUGS: Merged revisions 49045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) + | 2 lines location of the bug posting guidelines has changed + ........ + + * sample.call: simple commit to test CIA integration + +2006-12-28 21:26 +0000 [r49032-49035] Jason Parker + + * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me + + * main/http.c: saw this in passing... fix a small typo + +2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming + + * sounds/Makefile: new versions of sounds + +2006-12-28 19:52 +0000 [r49024] Jason Parker + + * main/http.c: make the uris_lock a rwlock instead of a mutex lock + - needs to be forward ported to trunk + +2006-12-28 19:43 +0000 [r49022] Joshua Colp + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Backport support for read/write locks. + +2006-12-28 19:21 +0000 [r49020] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, + pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, + pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, + pbx/ael/ael.tab.h, utils/ael_main.c: removed as in trunk + from the ael stuff. Also, threw in a minor fix to frame.c to + avoid build-killing compiler warnings. + +2006-12-27 22:28 +0000 [r49009] Joshua Colp + + * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not + available when LOW_MEMORY is used and things are being built in + the utils directory, so we need to resort to the old method of + strncpy. (issue #8579 reported by mottano) + +2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming + + * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, + main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, + main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, + main/http.c, main/logger.c: since these variables all have static + duration, none of them need initializers (they default to zero + anyway) + + * include/asterisk/options.h, main/asterisk.c, main/file.c: move + extern declaration for this option to a header file where it + belongs provide an initial value for 'languageprefix' option, + instead of relying on randomness to provide a useful value + +2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson + + * channels/chan_sip.c: Only include acl.h and lock.h once + + * channels/chan_sip.c: Only set rfc2833compensate flag once + (handle_common_options) + + * channels/chan_sip.c: - Remove checking for T38 options twice. + Keeping them in handle_common_options + +2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming + + * channels/chan_sip.c: make the option actually match the + documentation + + * channels/iax2-parser.c, include/asterisk/utils.h, + include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show + memory' and 'show memory summary' to distinguish memory + allocations that were done for caching purposes, so they don't + look like memory leaks + +2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more + politically correct + + * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy + cisco MWI support. Normally we try not to change our software for + bugs in other devices. But in this case, the Cisco phones are so + widespread so we try to implement a fix while waiting for a + bugfix from Cisco. + + * channels/chan_sip.c: - Make sure handle_common_options return 1 + when we found a common option - Move uncommon (only global) + option away from handle_common_options Reported by rizzo. Thanks! + + * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before + re-sending invite with auth. + + * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. + (rizzo, #8600) + +2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp + + * apps/app_meetme.c: Get rid of a needless memory allocation and + only create a conference structure in find_conf_realtime if data + was read from realtime. (issue #8669 reported by robl) + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an + API call that initializes an RTP structure. We need this because + chan_sip is cheeky and uses a temporary RTP structure for codec + purposes, and the API calls that are used rely on the lock. + (Pointed out on asterisk-dev by Andy Wang) + + * configure, configure.ac: Clean up autoconf file (gets rid of + warnings seen when rebuilding configure) and rebuild configure. + +2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant + + * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | + 6 lines Fix an error introduced by copying and pasting the + handling of the >= operator for the MATH function. If a single + equal sign was used as an operator, the function would treat it + is as if it were the >= operator. Now, it properly handles it as + an invalid operator. (issue #8665, patch by tempest1) ........ + + * channels/chan_oss.c: Fix a typo in an error message that + indicated that the MGCP channel type could not be registered, + instead of the correct type, OSS. + + * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | + 3 lines Check for the proper return value on an error in a call + to mmap(). This was reported by Andy Wang on the asterisk-dev + list. Thanks! ........ + + * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | + 3 lines Remove a couple of misplaced dots in log messages. This + was reported by Andrea Spadaccini on the asterisk-dev mailing + list. ........ + + * main/http.c: Implement locking for the list of URI handlers to + make it thread-safe. + +2006-12-23 Kevin P. Fleming + + * Asterisk 1.4.0 released. + +2006-12-22 22:33 +0000 [r48870-48906] Jason Parker + + * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. + + * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia + +2006-12-21 20:26 +0000 [r48783] Joshua Colp + + * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 + lines Add new silence sound files to the spec for Redhat. (issue + #8652 reported by alvaro_palma_aste) ........ + +2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp + + * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage + builds. + + * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so + it is then passed to the IMAP store file function. (issue #8614 + reported by punknow) + + * doc/snmp.txt: find is not the same as bind when it comes to + documentation. (issue #8626 reported by johann8384) + +2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming + + * channels/Makefile: suppress compiler warnings in this module + until it can be improved + +2006-12-19 21:12 +0000 [r48585] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 + lines Free localuser structure when we fail to dial (issue #8612 + reported by rizzo) ........ + +2006-12-19 21:03 +0000 [r48583] Luigi Rizzo + + * apps/app_sms.c: fix a bogus datalen in the frames generated by + app_sms (causing noisy output if you listen to the output!) This + affects trunk as well, whereas 1.2 is ok. + +2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming + + * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable + type for these unixODBC API calls, eliminating warnings on 64-bit + platforms that use the 'new' 64-bit types for ODBC API calls + +2006-12-19 03:46 +0000 [r48571] Joshua Colp + + * Makefile: Use env -i to start a fresh environment when going to + build menuselect. This is more portable then using unset. (issue + #8543 reported by jtodd) + +2006-12-18 17:23 +0000 [r48566] Luigi Rizzo + + * include/asterisk/channel.h: unbreak the macro used for + incrementing the frame counters. I don't know when the bug was + introduced, but with the typical usage c->fin = + FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects + trunk as well (fix coming). + +2006-12-18 17:15 +0000 [r48564] Joshua Colp + + * channels/chan_iax2.c: Put thread into proper list if we abort + handling due to an error, and also hold the lock while putting it + back into the proper idle list so we don't prematurely get a + signal. (issue #8604 reported by arkadia) + +2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming + + * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, + utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, + utils/ael_main.c: remove some now-unnecessary explicit includes + of autoconfig.h clean up per-file dependencies during 'make + clean' + + * build_tools/prep_tarball: need an additional argument here to + make the downloads actually occur + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep + these calls from thinking they have multiple arguments + + * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, + funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, + main, codecs/gsm, pbx, res, channels, codecs, utils, agi, + main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: + simplify dependency tracking system, using the compiler's + built-in method for generating them, and only doing dependency + tracking if developer mode is enabled via the configure script + + * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we + really, really have to have autoconfig.h included before all + other headers (especially system headers), the Makefile will now + force it to happen (this will fix build problems with files like + ast_expr2f.c, where we can't control the inclusion order in the + file itself) + + * funcs/func_curl.c: instead of initializing the curl library every + time the CURL() function is invoked, do it only once per thread + (this allows multiple calls to CURL() in the dialplan for a + channel to run much more quickly, and also to re-use connections + to the server) (thanks to JerJer for frequently complaining about + this performance problem) + +2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp + + * main/rtp.c: Turn payload_lock into bridge_lock and make it + encompass all RTP structure contents that may relate to bridge + information, including who we are bridged to. + + * channels/chan_iax2.c: Hold call structure lock in places where a + qualify or peer action can destroy it. + + * channels/chan_iax2.c: Lock network retransmission queue in all + places that it is used. + +2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson + + * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported + from 1.2) + + * channels/chan_sip.c: Update to latest IANA spec + +2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp + + * channels/chan_iax2.c: Use a wakeup variable so that we don't wait + on IO indefinitely if packets need to be retransmitted. + + * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP + structure can change AFTER a bridge has started. This comes from + the packet handling of the SIP response when indication that it + was answered has been sent. Therefore we need to protect this + data with a lock when we read/write. (issue #8232 reported by + tgrman) + + * main/rtp.c: Remove direct RTCP bridging. I've come to the + conclusion that we should handle this through the core and not + just forward it on. Should solve a few bugs. + +2006-12-12 Kevin P. Fleming + + * Asterisk 1.4.0-beta4 released. + +2006-12-12 04:13 +0000 [r48401] Joshua Colp + + * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This + is the way it should have been done. + +2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman + + * sounds/Makefile: new sounds package with 100% more silence + + * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge + from https://svn.digium.com/svn/asterisk/branches/1.2 ........ + r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) + | 4 lines app_externalivr needs a real silence file, and + additional changes to add silence files into core instead of + extra patch provided by bug 8177 with minor additions. ........ + +2006-12-11 21:31 +0000 [r48391] Joshua Colp + + * apps/app_voicemail.c: Return non-existant callerid handling to + that which it was before. In 1.4 and trunk callerid can be + allocated but not have any contents so we have to use + ast_strlen_zero before passing it to the relevant functions. + (issue #8567 reported by pabelanger) + +2006-12-11 05:37 +0000 [r48382] Tilghman Lesher + + * funcs/func_strings.c: STRFTIME() does not actually require an + argument (issue 8540) + +2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp + + * main/rtp.c: Merge in my latest RTP changes. Break out RTP and + RTCP callback functions so they no longer share a common one. + + * apps/app_meetme.c: Use the correct API call to say a device state + changed. (Yes, I'm a nub.) + + * apps/app_meetme.c: Don't access the conference structure after it + has been freed. + +2006-12-11 00:47 +0000 [r48375] Tilghman Lesher + + * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, + res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, + apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) + | 5 lines When doing a fork() and exec(), two problems existed + (Issue 8086): 1) Ignored signals stayed ignored after the exec(). + 2) Signals could possibly fire between the fork() and exec(), + causing Asterisk signal handlers within the child to execute, + which caused nasty race conditions. ........ + +2006-12-10 03:04 +0000 [r48372] Steve Murphy + + * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 + line This version applies the patch suggested by stevens in bug + 7836 (make inbound channel RINGING state consistent with other + channels). ........ + +2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant + + * channels/chan_iax2.c: Use locking when accessing the + registrations list. This list is not actually used very often, so + the likelihood of there being a problem is pretty small, but + still possible. For example, if the CLI command to list the + registrations was called at the same time that a reload was + occurring and the registrations list was getting destroyed and + rebuilt, a crash could occur. In passing, go ahead and convert + this list to use the linked list macros. + +2006-12-07 18:17 +0000 [r48357] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 + Dec 2006) | 3 lines Ensure that the file position is not + incremented beyond the total number of files available for + playback. (issue #8539, ulogic) ........ + +2006-12-07 15:33 +0000 [r48349] Steve Murphy + + * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that + killed bug 8423 -- OriginateSuccess and OriginateError incomplete + channel name. May it rest in peace. + +2006-12-06 16:25 +0000 [r48326] Olle Johansson + + * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being + retransmitted to Asterisk + +2006-12-06 16:15 +0000 [r48323] Russell Bryant + + * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 + Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option + in the sample configuration file. (issue #8526, arkadia) ........ + +2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson + + * /, channels/chan_sip.c: Don't send Contact on MESSAGE + +2006-12-05 20:42 +0000 [r48279] Jason Parker + + * configure.ac: Fix curl version number testing to be much more + friendly to non-bash shells. Issue 8508, patch by me. This + *SHOULD* be POSIX compliant now.. + +2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson + + * channels/chan_sip.c: Merging the invitestate-1.4 branch after + successful testing. Will check if I can solve this with less + changes in 1.2. + + * configs/sip.conf.sample: Add missing s from another repository. + (thanks jcmoore!) + + * configs/sip.conf.sample: Updating sip.conf.sample with + information about T38 not working when chan_local or chan_agent + is involved in the call. I don't know how big a fix that would be + to solve, but this is the current state of affairs. (Chan_sip + currently checks if the other side of the bridge has a SIP tech. + We could/should implement another check, possibly for udptl_write + or some flag in the ast_channel structure). + +2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher + + * apps/app_voicemail.c: Oops, forgot to release the odbc handle + + * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) + | 6 lines If the recording in the database is too large, it will + fail to retrieve with an mmap error. Not too sure why this + doesn't happen when we put it in the database, also, but since + that doesn't seem to be broken, I'm not going to fix it (at least + until someone reports it). Solution is to ask for the file in + smaller chunks. (Bug 8385) ........ + +2006-12-04 21:48 +0000 [r48237-48248] Jason Parker + + * apps/app_voicemail.c: Fix an issue which didn't allow + unavail/greet/busy/etc messages from being saved into ODBC (and + probably IMAP). + +2006-12-04 17:54 +0000 [r48228-48230] Jason Parker + + * configs/voicemail.conf.sample: Add documentation to + voicemail.conf.sample for ODBC storage. Issue 8499 - patch by + blitzrage. + + * doc/snmp.txt: Attempt to document some of the dependencies that + are needed for net-snmp Issue 8499 - initial patch by blitzrage. + +2006-12-03 06:34 +0000 [r48223] Russell Bryant + + * sounds/Makefile: When "fetch" is in use, instead of "wget", + --continue is not a valid option. (issue #8451) + +2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson + + * channels/chan_sip.c: - Removing one of two pieces of code to + handle 481 response on INVITE - Move handling of REFER response + to handle_response_refer() + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, + configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax + transmission happens - Encapsulate RTP timers in the rtp + structure so we have one for video and one for audio The video + one is not used in 1.4, really. Will be used for RTP keepalives + when we can send something that video phones support in the RTP + stream. I now this is a big architectual change at this stage for + 1.4, but decided it was needed to avoid future bug reports. - + Document the RTP NAT keepalive option in sip.conf.sample Issue + 7679 in the bug tracker. Please test. + +2006-12-02 03:50 +0000 [r48195] Russell Bryant + + * include/asterisk/utils.h: Backport the comment containing the + warning regarding the limitations on the usage of this function. + It is thread safe, but not technically reentrant. + +2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming + + * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) + | 2 lines if Dial() is going to send music-on-hold to the calling + party, it has to send PROGRESS first to ensure that the reverse + audio path has been setup first (BE-106) ........ + +2006-12-01 23:16 +0000 [r48190] Russell Bryant + + * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: + FreeBSD 6.1 does not include wget by default. However, it has + fetch which will work just fine for our purposes of downloading + the sounds packages. So, check for both wget and fetch and the + configure script and use what was found to download them. If + neither one was found, and sound packages are selected that must + be downloaded, the install process will print out an informative + error message indicating the situation. Also, fix a couple places + where "make" was hard coded into some output messages by + replacing them with the $(MAKE) variable. (issue #8451, initial + patch by pabelanger, with additional modifications by me) + +2006-12-01 20:25 +0000 [r48184-48186] Jason Parker + + * configs/extensions.conf.sample, /: Merged revisions 48183 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 + lines Fix a small typo - issue 8848, reported by pabelanger + ........ + +2006-12-01 19:38 +0000 [r48179] Tilghman Lesher + + * main/cli.c: Double-unlock error (reported by blitzrage on IRC) + +2006-12-01 17:41 +0000 [r48177] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the + "limitonpeers" patch from trunk, to fix a lot of issues with + queues and SIP device states - Remove support for T.38 early + media, since it's impossible. (Two patches in one - extra friday + evening offer due to being off line from svn today... :-) + +2006-11-30 21:18 +0000 [r48168] Joshua Colp + + * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not + do a partial bridge for Google Talk since we need to handle STUN. + (issue #8448 reported by phsultan) + +2006-11-30 20:51 +0000 [r48166] Olle Johansson + + * /, channels/chan_sip.c: Issue 8319 - change noncecount before + using it. + +2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp + + * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 + lines Only print out debug message if bridged channel is not + NULL. (issue #8412 reported by jubilex) ........ + + * /, res/res_features.c: Merged revisions 48154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 + lines Do not listen for DTMF on the bridge that comes into + existence when ParkedCall is executed. This means native bridging + can now occur for this. (issue #8406 reported by kebl0155) + ........ + + * main/cdr.c, /: Merged revisions 48151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 + lines Print certain CDR messages out at the NOTICE level versus + WARNING since they can occur when used with the CDR applications + and are perfectly fine. (issue #8367 reported by dartvader) + ........ + + * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov + 2006) | 2 lines Document 'port' for SIP peers, came up because of + the current mailing list thread. (issue #8450 reported by + blitzrage) ........ + +2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson + + * doc/manager.txt: Explain status reports and make codefreeze more + happy :-) + + * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by + GS 487 adapter without CSEQ on separate line in the REGISTER + request. Imported from 1.2. + +2006-11-29 21:05 +0000 [r48115] Joshua Colp + + * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in + mm_login. (issue #8420 reported by slimey) + +2006-11-29 19:56 +0000 [r48113] Olle Johansson + + * configs/sip.conf.sample: Explain the use device status system + implemented in SIP for subscriptions, queues and manager a bit + better. Like in 1.2, you will get more detailed information if + you set a call limit for a device. When the call limit is + reached, the status system will report a device as busy. For + queues, setting a call limit per SIP device is propably a + requirement. In most cases, it will work much better if you only + use type=peer and not type=friend. We might decide to backport + the new setting from trunk to apply all call limits to the peer + part of a friend only. + +2006-11-29 16:50 +0000 [r48107] Joshua Colp + + * main/rtp.c, /: Merged revisions 48106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 + lines If the frame was duplicated before writing out then we need + to free it. (issue #8429 reported by edguy3) ........ + +2006-11-29 08:03 +0000 [r48105] Olle Johansson + + * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. + +2006-11-29 04:26 +0000 [r48101] Joshua Colp + + * apps/app_voicemail.c: Don't crash if the mailstream was not + created. + +2006-11-28 18:26 +0000 [r48095] Jason Parker + + * Makefile: Export several more variables in top level Makefile. + Inspired by issue 8438. + +2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp + + * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov + 2006) | 2 lines According to the research I have done we never + needed to include compiler.h in the first place so let's not! + (issue #8430 reported by edguy3) ........ + + * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 + lines Use the proper function to get the new message count + instead of always using the filesystem. (issue #8421 reported by + slimey) ........ + +2006-11-27 17:20 +0000 [r48049] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 + Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) + ........ + +2006-11-27 17:17 +0000 [r48046] Russell Bryant + + * main/manager.c: Remove a couple of unused variables (issue #8380, + casper) + +2006-11-27 15:32 +0000 [r48038] Joshua Colp + + * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 + lines Do not reference the freed outgoing structure in the debug + message. (issue #8425 reported by arkadia) ........ + +2006-11-27 06:41 +0000 [r48031] Olle Johansson + + * channels/chan_sip.c: Change logging message + +2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy + + * funcs/func_cdr.c: might as well also document the raw values of + the flag vars + + * /, funcs/func_cdr.c: A little bit of func_cdr documentation + upgrade-- no bug# involved, although 8221 may have inspired it. + +2006-11-25 09:28 +0000 [r48002] Olle Johansson + + * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 + and future releases, you can disable subscription support totally + or per peer in sip.conf with allowsubscribe = yes | no + +2006-11-24 17:17 +0000 [r47992] Steve Murphy + + * main/translate.c: bug 8189 posted this fix for main/translate.c + for PLC + +2006-11-24 15:46 +0000 [r47989] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 + Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. + beatufied some logs, changed some loglevels. changed the default + value of block_on_alarm ........ + +2006-11-23 11:01 +0000 [r47959] Olle Johansson + + * /, channels/chan_sip.c: Don't allocate unused variable. + +2006-11-22 21:47 +0000 [r47944] Joshua Colp + + * main/rtp.c: Video will never reach Packet2Packet bridging and can + do more harm then good. + +2006-11-21 17:32 +0000 [r47897] Joshua Colp + + * main/rtp.c: If we have the non standard G726-32 setting turned on + we want to return G726-32 to the SDP, not our AAL2 string. (issue + #8330 reported by voipgate) + +2006-11-21 15:20 +0000 [r47892] Olle Johansson + + * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 + provisional response. Let's not treat that as early media. + (discovered at the AVTF meeting in Paris). + +2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher + + * apps/app_voicemail.c: Oops, merge missed release of odbc object + + * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) + | 2 lines Failing to trap -1 error from mmap causes segfault + (Issue 8385) ........ + +2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp + + * main/frame.c, /: Merged revisions 47859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 + lines Don't forget to byte swap if we are exiting the smoother + feed early. (issue #8287 reported by arturs) ........ + +2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming + + * /, doc/billing.txt: update documentation regarding IAX2 transfers + and CDRs Merged revisions 47776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) + | 2 lines update clearly wrong documentation regarding cdr_custom + ........ + +2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp + + * channels/chan_sip.c: Compare technology using the pointers + instead of a straight comparison based on name. (issue #8228 + reported by dean bath) + +2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming + + * configure, configure.ac: check for pre-1.4 versions of Zaptel and + abort the configure script if found with an appropriate error + message + +2006-11-16 19:24 +0000 [r47755] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD + notification optional, in order to avoid a lot of extra database + lookups for all those realtime users out there. + +2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 47750 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov + 2006) | 2 lines Because of the way chan_local is written we + should be extra careful and make sure our callback functions have + a tech_pvt. (issue #8275 reported by mflorell) ........ + + * apps/app_meetme.c: Don't unreference the SLA object if there is + no SLA object in the devicestate callback. (issue #8354 reported + by loloski) + +2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson + + * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup + + * UPGRADE.txt: Warn users about change in canreinvite + + * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never + authenticated (according to the RFC) - Update docs on + canreinvite. "nonat" is the recommended setting for most users + with phones behind a NAT. + +2006-11-15 22:31 +0000 [r47712] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 47711 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov + 2006) | 2 lines Make sure that the pvt structure exists before + trying to do fixup on Local channels. (issue #7937 reported by + mada123, fix by alamantia with mods by me) ........ + +2006-11-15 21:56 +0000 [r47709] Tilghman Lesher + + * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL + +2006-11-15 21:33 +0000 [r47707] Joshua Colp + + * main/channel.c: We need to ensure timelimit stuff is included as + well so warnings get played. (issue #8050 reported by KNK) + +2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming + + * main/file.c: don't try to call fclose() if fopen() failed + +2006-11-15 20:31 +0000 [r47698] Olle Johansson + + * channels/chan_sip.c: - Improve SIP history - Never send reply to + ACK (again...) + +2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming + + * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) + | 4 lines ensure that message duration is included in email + notifications for forwarded messages (BE-96, fix by me after + corydon used his clue-bat on me) ensure that duration in the + message metadata is updated if prepending is done during + forwarding (related to BE-96) remove prototype for API call that + does not exist ........ + + * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 + Nov 2006) | 2 lines clear the category's variable tail pointer as + well when variables are detached from it ........ r47688 | + kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 + lines when appending a list of variable to a category, ensure the + tail pointer points to the last variable in the list ........ + r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) + | 2 lines when re-writing the config file, don't repeat the path + if it hasn't changed ........ + + * main/config.c, /: Merged revisions 47682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) + | 2 lines ouch... don't use printf, use ast_log/ast_verbose + ........ + +2006-11-15 17:46 +0000 [r47672] Luigi Rizzo + + * main/cli.c: fix longest match search in find_cli. Trunk already + fixed. 1.2 not affected (well, i have no idea, the code is + totally different there). + +2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson + + * /, channels/chan_sip.c: Send error message when we can't allocate + SIP dialog, possibly due to limitation of file descriptors. + (imported from 1.2) + +2006-11-15 04:45 +0000 [r47645] Joshua Colp + + * main/rtp.c: If NAT detection is turned on or already detected + then say NAT is active when setting the remote RTP peer when + doing early bridging. (issue #8365 reported by marcelbarbulescu) + +2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming + + * main/term.c: more formatting cleanup, and avoid running off the + end of the string + +2006-11-15 00:14 +0000 [r47639] Joshua Colp + + * main/rtp.c: Turn notice about unknown RTCP packet type into a + debug message instead. + +2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming + + * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit + platforms (this variable is an 'int' anyway, comparing it to + 'signed long' is not useful) + +2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 + lines Update copyright information in the ADSI logo blob. + ........ + + * channels/chan_sip.c: Only keep the video RTP structure around if + 1. Video support is enabled and 2. A video codec is enabled on + the dialog + + * funcs/func_uri.c: Small documentation clarification for + URIENCODE. (issue #8294 reported by salaud) + +2006-11-14 18:54 +0000 [r47621] Tilghman Lesher + + * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ + prefix did not completely transition app_voicemail when + ODBC_STORAGE is used (reported on IRC by caio1982, not in + bugtracker) + +2006-11-14 16:45 +0000 [r47617] Joshua Colp + + * apps/app_amd.c: Use LOG_DEBUG to print out the indication that + app_amd is using default settings instead of using LOG_NOTICE. + This stops needless logging of this information under normal + circumstances. (issue #8361 reported by Seb7) + +2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson + + * channels/chan_sip.c: Update documentation to fit the + implementation... + + * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in + retransmission system if it's an OPTION packet from peerpoke + +2006-11-13 21:28 +0000 [r47584] Joshua Colp + + * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 + lines Initialize global pointers for connection and result to + NULL. (issue #8356 reported by james) ........ + +2006-11-13 20:20 +0000 [r47581] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) + | 2 lines Having more than 255 old messages caused corruption in + the new/old count ........ + +2006-11-13 19:15 +0000 [r47576] Steve Murphy + + * main/config.c: This solves bug 8342, whereby a crash occurs under + certain circumstances while reading a config file with comments-- + a call to CB_ADD shouldn't happen if withcomments is zero + +2006-11-13 19:11 +0000 [r47573] Tilghman Lesher + + * main/cli.c, channels/chan_sip.c: Re-enable old deprecated + commands + +2006-11-13 19:10 +0000 [r47572] Olle Johansson + + * /, channels/chan_sip.c: - Don't reply to INVITE already replied + to when we get BYE - Declare errmsg as int. Oops. + +2006-11-13 18:18 +0000 [r47564] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing + the messed if, but we all forgot to update the regressions. Until + now. + +2006-11-13 17:13 +0000 [r47553] Steve Murphy + + * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being + found... just confuses users + +2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp + + * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 + lines When sending an SMS with a user data header properly set + the UDH flag in the first byte. (issue #8347 reported by + hoffmeis) ........ + + * main/cli.c: Free full command string upon unregistering of CLI + command. Backported from revision 47536 from rizzo. + +2006-11-13 16:00 +0000 [r47540] Olle Johansson + + * channels/chan_sip.c: Only produce error message about sip history + once + +2006-11-13 05:48 +0000 [r47527] Russell Bryant + + * configure, acinclude.m4: AC_PROG_SED is included in autoconf + 2.60, but apparently it is not included in 2.59. So, to maintain + compatability with 2.59 since it is a small change, copy this + macro into acinclude.m4 and rename it to AST_PROG_SED. (issue + #8345) + +2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher + + * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) + | 2 lines If the execute fails a second time, make sure that we + don't pass back a stale handle ........ + + * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) + | 2 lines Don't play dialtone if the seizing the channel fails + (Bug 7754) ........ + +2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson + + * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks + DEA!!!) + + * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is + UDPTL in sdp... + + * channels/chan_sip.c: - Don't destroy SIP dialog because of a + failed T.38 re-invite. Wait for a bye. Final response to a + re-invite does not mean that the session dies, only that the + re-invite fails. - Keep RTP active during processing of T.38 + re-invite. If the re-invite fails, RTP needs to remain as before + the re-invite. Issue 8338 - darren1713. Please test. + + * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp + -Add some comments to t.38 code + +2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | + 4 lines Only do the check to determine whether the channel + calling this function is an IAX2 channel when getting the IP + address using the special argument, CURRENTCHANNEL. (issue #8341, + jcovert) ........ + + * Makefile: Add the target "menuconfig" as an alias for the + "menuselect" target. This is just a favor to users so that if you + accidentally type "make menuconfig" instead of "make menuselect", + it still works. (inspired by a comment on IRC from wangster + calling me an "especially devious asterisk developer" for having + it be menuselect instead of menuconfig. :) ) + + * main/term.c: Tweak the formatting of this new function to better + conform to coding guidelines. + +2006-11-11 02:04 +0000 [r47490] Matt O'Gorman + + * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo + safe output! + +2006-11-10 22:23 +0000 [r47480] Matt Frederickson + + * channels/chan_zap.c: Make sure we don't use 32 bits when we only + need one bit. + +2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson + + * channels/chan_sip.c: ...and make sure that the dialog is + destroyed, even if we don't get any answer on the bye... This is + the channel that remains dead after the SIP transfer + + * channels/chan_sip.c: Add debug output while trying to trace bug + in bug report + + * channels/chan_sip.c: Make sure we destroy dialog... + + * /, channels/chan_sip.c: Small cleanup of handle_request_invite() + - imported from 1.2 with changes + +2006-11-10 19:47 +0000 [r47462] Matt Frederickson + + * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide + callerid name for switches that bork on it. + +2006-11-10 18:56 +0000 [r47454] Olle Johansson + + * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart + SDP (alphaque) + +2006-11-10 17:13 +0000 [r47444] Luigi Rizzo + + * build_tools/prep_moduledeps: grep -m is not available on BSD, so + use head -1 instead + +2006-11-10 16:53 +0000 [r47437] Joshua Colp + + * apps/app_chanspy.c: Only split up extension and context if a + value exists. (issue #8332 reported by loloski) + +2006-11-10 16:51 +0000 [r47436] Tilghman Lesher + + * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_iax2.c: Discussion of these CLI changes resulted in + more consistency (Bug 8236) + +2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming + + * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then + removing them should be LOG_NOTICE, not LOG_DEBUG + + * apps/app_queue.c: reflect addition/removal of dynamic queue + members in queue_log, so that people using dialplan replacement + for AgentCallbackLogin can still track login/logout (issue #7736, + reported/patched by whoiswes but this commit was written by me + and covers all three paths for AQM/RQM) + +2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson + + * channels/chan_sip.c: Rip out half implementation of 491 response + support, since it wasn't implemented properly and caused memory + leaks in the case of us getting 491's, which Asterisk actually + sends... Since it is a bit too complicated to fix this, I'll rip + it out of 1.4 and put it on the to-do-list for future releases. + Now, we handle this as congestion, which it really is. Issue + #8331 + + * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. + Thanks fenlander! + +2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp + + * channels/chan_h323.c: Fix building of chan_h323 by completeing + some structure definitions. (issue #8327 reported by Mithraen) + + * apps/app_voicemail.c: Do conversion in a more easier to read and + working way for \r, \n, and \t. (issue #8324 reported by + johnlange) + +2006-11-09 21:26 +0000 [r47391] Russell Bryant + + * apps/app_voicemail.c, channels/chan_zap.c, + build_tools/prep_moduledeps: Work around an issue that caused + menuselect to display a bogus description for app_voicemail and + chan_zap. These modules use some preprocessor directives to + determine what it will report to Asterisk as its description. + However, the way we extract this information from the source + files for menuselect is not smart enough to figure this out. + (issue #8326, #8328) + +2006-11-09 16:53 +0000 [r47380] Joshua Colp + + * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov + 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and + higher as, well, it's apparently going to be removed. This should + make all you FC6 fans happy as your Asterisk will now build + without any mods. ........ + +2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant + + * main/cli.c: fix tab completion for "core debug channel" and "core + no debug channel" + + * main/cli.c: Fix "core show channel". Also, fix tab completion for + both "core show channel" and "core show channels". + + * main/cli.c: Fix "core debug channel ". I guess someone + needs to go through and audit every CLI command that changed + number of arguments ... + + * main/asterisk.c: revert the previous change, which actually + modified the deprecated command, "show profile". Now, actually + apply the change to "core show profile". + + * main/asterisk.c: Fix argument parsing for the "core show profile" + CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) + + * main/cli.c: Fix another CLI command, "core show uptime" ... + (issue #8323, reported by johnlange, fixed by myself) + + * main/asterisk.c: fix "core show version" to reflect the new + number of arguments for this CLI command (issue #8316, kshumard) + +2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy + + * main/channel.c: This update fixes 7531 + + * channels/chan_skinny.c: Committed in behalf of 8190. + +2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming + + * main/frame.c: the battle over CLI command formats has broken + stuff... + + * channels/chan_sip.c: add simple fix for SDP to report proper + sample rate for G.722 media sessions + +2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant + + * utils/streamplayer.c: I occasionally get email from users that + are trying to figure out what this does, or due to some + misunderstanding as to what it is supposed to do, can't get it to + work. So, I have added some text here to hopefully explain what + this application does and does not do. + + * channels/chan_gtalk.c: Make this module build again + + * configure, configure.ac, acinclude.m4: Copy the macros from + libtool.m4 to our own acinclude.m4 such that libtool is no longer + required to be installed to be able to generated the configure + script. + +2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson + + * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) + +2006-11-07 23:46 +0000 [r47303] Steve Murphy + + * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, + channels/chan_misdn.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_h323.c, + channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, + include/asterisk/stringfields.h, apps/app_voicemail.c, + main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, + channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, + channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to + solve the problem in bug 7506. It's a lot of rework to solve a + fairly small problem... such is life. + +2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp + + * channels/chan_local.c: Make MOH work as it did before in + chan_local, without this then it can go funky when transfers and + MOH are involved. (issue #7671 reported by jmls) + +2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming + + * configs/musiconhold.conf.sample: clean up sample config, and make + native file playback the more obvious default choice + +2006-11-07 18:38 +0000 [r47275] Matt O'Gorman + + * apps/app_voicemail.c: large overhaul to voicemail imap support. + Allows support for more imap servers, also a better + implementation of several parts of the original work. patch + provided by 8033 with major upgrades. + +2006-11-07 17:30 +0000 [r47268] Olle Johansson + + * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of + continue. + +2006-11-07 13:13 +0000 [r47250] Olle Johansson + + * /, channels/chan_sip.c: Fixing the attack shield so it doesn't + produce attacks... Issue 8265 - never reply to an ACK + +2006-11-07 01:25 +0000 [r47239] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 + Nov 2006) | 5 lines If random order is enabled for files mode + music on hold, set a random initial position, instead of always + starting at the first file, and doing the random operation only + when switching to the next file. (bug reported by John Lange on + the asterisk-dev mailing list) ........ + +2006-11-04 18:32 +0000 [r47199] Olle Johansson + + * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and + transfer from "john" Thank you! + +2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant + + * main/cli.c: Fix another bug in "core set debug" ... + + * main/asterisk.c, main/cli.c: Really fix the "core set debug" and + "core set verbose" CLI commands. + + * main/cli.c: fix the "atleast" option to the "core set verbose" + and "core set debug" CLI commands + +2006-11-03 23:17 +0000 [r47176] Steve Murphy + + * channels/chan_sip.c: This fix introduced via bug 8233 + +2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo + + * bootstrap.sh: align bootstrap.sh with the version in trunk (needs + to be blocked as it is already in trunk) + + * configure.ac: add proper environment vars to detect modules on + freebsd. (already applied to trunk so it needs to be blocked + there) + +2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher + + * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, + channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More + changes making the CLI more consistent with "category verb + arguments" (continuation of issue 8236) + + * main/config.c, main/cli.c, main/channel.c, main/manager.c, + channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, + main/http.c, main/file.c, main/logger.c, main/image.c, + res/res_indications.c, main/asterisk.c, res/res_odbc.c, + channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_local.c, main/frame.c, channels/chan_sip.c, + res/res_features.c, channels/chan_agent.c, res/res_crypto.c, + res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: + Reverse change of "show" to "list" and make several other + commands more consistent with "category verb arguments" + +2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson + + * channels/chan_sip.c: Move check for codec translation to + sip_call() instead of in add_sdp. No one bothers with the result + of add_sdp anyway... Yet... + + * channels/chan_sip.c: Disable code for T38 over TCP and RTP since + there's no trace of actual functionality for it :-) + +2006-11-02 17:49 +0000 [r46965] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 + Nov 2006) | 3 lines ignore files in a music on hold directory + that begin with '.' (issue #8249, cboie) ........ + +2006-11-02 17:17 +0000 [r46963] Nadi Sarrar + + * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix + +2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming + + * channels/chan_sip.c: don't send INVITE when we have determined + that we can't offer any audio formats due to lack of transcoding + support (or incorrect configuration) + +2006-11-02 16:06 +0000 [r46930] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 + lines Repeat after me oej: I will at least make sure my code + compiles before I commit it. ........ + +2006-11-02 15:24 +0000 [r46901] Olle Johansson + + * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) + +2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant + + * /, main/callerid.c: Add the missing call to free described in + issue #8268. Also, add a bunch of missing calls to free in + callerid_feed_jp(). + + * main/say.c: fix saying one hundred and two hundred in hebrew + (issue #7810, eldadran) + + * Makefile, configure, codecs/gsm/Makefile, configure.ac, + build_tools/strip_nonapi, makeopts.in: Fixes for + cross-compilation on mips (issue #8058, ywalther, with some + modifications) + + * aclocal.m4, build_tools/menuselect-deps.in, configure, + build_tools/embed_modules.xml, configure.ac: Add a check in the + configure script to determine whether ld is GNU ld or not. This + is needed because module embedding only works for gnu ld. GNU ld + is now listed as a dependency for all of the module embedding + options in menuselect. (issue #8143) + +2006-11-01 20:35 +0000 [r46822] Matt O'Gorman + + * channels/chan_gtalk.c: bind address support from bug 8164 + +2006-11-01 19:49 +0000 [r46802] Steve Murphy + + * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to + accept longer strings or mass confusion and a lot of lost time is + the result + +2006-11-01 18:39 +0000 [r46780] Joshua Colp + + * main/Makefile: Force poll() emulation for Darwin to always be on. + It's too broken to consider being used. This resolves the console + issue OSX users have been seeing. I would have liked to autoconf + this but I haven't been able to come up with a test case that + works. Que sera. + +2006-11-01 18:26 +0000 [r46778] Russell Bryant + + * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | + 9 lines soxmix and Asterisk expect different file extensions for + certain formats. This was already handled for the wav49 format. + However, it was not handled for ulaw and alaw. I fixed this in + such a way that using the alternate extensions for ulaw and alaw + will only happen if we know we're calling soxmix, and not a + custom script defined using the MONITOR_EXEC variable. The wav49 + processing was left alone so that external scripts will see no + behavior change. (issue #7550, reported by mnicholson, proposed + patch by junky, committed fix is a bit different) ........ + +2006-11-01 18:21 +0000 [r46775] Joshua Colp + + * channels/chan_iax2.c: It's another round of chan_iax2 fixes! + Should hopefully fix the deadlock issues people have been + reporting. IAXtel now has qualify turned on for 800 peers and it + is handling it fine. + +2006-11-01 17:48 +0000 [r46760] Steve Murphy + + * main/config.c: Cleanups suggested by Russell. + +2006-11-01 16:39 +0000 [r46744] Russell Bryant + + * channels/chan_zap.c: Prevent an infinite loop when config + processing gets to a jitterbuffer option + +2006-10-31 22:02 +0000 [r46716] Jason Parker + + * main/translate.c: Fix "core show translation" output. Issue + #8243, patch by Damin. + +2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming + + * include/asterisk/translate.h, main/translate.c: add an API so + that translators can activate/deactivate themselves when needed + + * include/asterisk/translate.h, main/translate.c: revert changes + that were the wrong way to address this... proper fix coming + + * main/translate.c: let's set the seen flag early enough to + actually make a difference... + + * include/asterisk/translate.h, main/translate.c: don't re-do setup + operations for translators that can dynamically register + themselves + +2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson + + * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue + #8089 - Fix the ENUM support (picking one record by number). + Thanks otmar! + + * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport + when we're supposed to support ;rport. Issue #7473. + + * /, channels/chan_sip.c: If peer fails ACL check, fail peer at + REGISTER + + * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! + +2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant + + * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the + boot process to ensure it starts after stuff like MySQL (issue + #8253, Alric) + + * /, main/utils.c: Merged revisions 46560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | + 3 lines When handling the case where the hostname is just an IPV4 + numeric address, be sure to set the address type. (issue #8247, + alexr) ........ + + * /, res/res_agi.c: Merged revisions 46557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | + 3 lines fix some copy/paste bugs in the checking of arguments for + the "control stream file" AGI command (issue #8255, mnicholson) + ........ + + * main/translate.c: Add a small tweak to the code that checks to + see whether destination formats are translatable based on the + source format. If we have already determined that there is no + translation path in one direction, don't bother checking the + other direction. + +2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming + + * main/translate.c: when unregistering a translator, don't rebuild + the translation matrix unless needed when filtering formats out + of an offer, ensure we check for translation ability in both + directions + + * include/asterisk/linkedlists.h: ensure that items removed from a + list are always unlinked from the list (next pointer set to NULL) + +2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp + + * configure, configure.ac: Don't explicitly link in crypt as it is + not used on some platforms. + + * channels/chan_iax2.c: We need to lock the pvt structure during + retransmission as another worker thread may be doing something as + well. + +2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson + + * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, + include/asterisk/doxyref.h, channels/chan_sip.c, + main/ast_expr2f.c, include/asterisk/module.h, + formats/format_ogg_vorbis.c, main/app.c, + include/asterisk/channel.h, include/asterisk/lock.h, + include/asterisk/frame.h: Issue #8246 - Doxygen fixes from + kshumard. An extra big thankyou is given to everyone that + contributes to doxygen! THANK YOU! + + * main/rtp.c, /: Bind RTCP to the same IP as RTP + + * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 + redirects (imported from 1.2) + + * /, channels/chan_sip.c: Issue #7608 - Notifications sent with + wrong content-type (imported from 1.2, modified) + + * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that + was reported for trunk, but obviously exists in 1.4 too. + + * channels/chan_sip.c: Restoring the old logic, since working + around it and fixing it seemed too complicated. - The + SIP_OUTGOING flag indicates the direction of the last transaction + in the dialog. - The initreq stores the last request in the + dialog, the request that opened the latest transaction. Please + now retry all the 1.4 bug reports with mixed to/from headers, + tags etc in ACK, BYE, CANCEL. Thanks! + + * channels/chan_sip.c: Accepting a message twice may be + misinterpreted... + + * channels/chan_sip.c: - 183 is not reliable message... - Error + should not have SDP + +2006-10-28 16:37 +0000 [r46377] Joshua Colp + + * utils/Makefile: Don't build muted on OpenBSD, it is not + supported. + +2006-10-27 19:03 +0000 [r46370] Russell Bryant + + * channels/chan_zap.c: move the copy of the default settings to the + global settings back out of process_zap, so that they aren't + overwritten when process_zap is called multiple times + +2006-10-27 18:29 +0000 [r46367] Olle Johansson + + * contrib/asterisk-ng-doxygen: Put some doxygen pressure on + Christian :-) + +2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant + + * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, + res/res_musiconhold.c: We should always be using _exit() after a + fork() or vfork() instead of exit(). This is because exit() does + some extra cleanup which in some implementations of vfork(), for + example, can actually modify the state of the parent process, + causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) + + * channels/chan_zap.c: Instead of iterating all of the options once + to look for jitterbuffer options, and then again for everything + else, move the processing of jitterbuffer options into the main + loop so that there are no erroneous messages about ignoring + unknown options. (issue #8226) + +2006-10-27 10:03 +0000 [r46351-46353] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 46350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | + 1 line fixed a bug which caused chan_misdn to try to allocate 2 + times the same channel on high load, which then caused + instability of mISDN. removed a useless function from isdn_lib.c + ........ + + * channels/misdn_config.c: fixed not compile issue, which was just + introduced + + * channels/misdn_config.c, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + Merged revisions 46176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | + 1 line added nttimeout option to configure wether we disconnect + calls on NT timeouts or not during an overlapdial session + ........ + +2006-10-26 17:57 +0000 [r46335-46340] Jason Parker + + * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 + lines oops - somebody forgot to change this - long ago, probably. + ........ + + * CHANGES: grammar check + +2006-10-26 16:38 +0000 [r46331] Olle Johansson + + * CHANGES: Corrections to changes (Multiparking is not included) + +2006-10-26 16:31 +0000 [r46329] Russell Bryant + + * main/translate.c: - If the source has no audio or no video + portion, do not call powerof() to get the format index. - Don't + run through the audio and video loops if there is no audio or + video portion of the source If 0 is passed to powerof, it will + return -1. This value of -1 was then being used as an array index + in these loops, which caused a crash on some systems. Other than + this issue, this code works as we expected it to. If a format is + not in the source, and we have to translation path to it, it is + not offered in the list of acceptable destination formats. (fixes + issue #8231) + +2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming + + * CHANGES: update to reflect G.722 addition + +2006-10-26 04:18 +0000 [r46298] Russell Bryant + + * doc/backtrace.txt: update backtrace documentation to reflect + changes in 1.4 (issue #8230, kshumard) + +2006-10-26 01:37 +0000 [r46287] Mark Spencer + + * main/config.c, main/manager.c: Fix config comment code + preservation code (thanks murf!) + +2006-10-25 20:14 +0000 [r46276] Olle Johansson + + * channels/chan_sip.c: Old todo note - Don't add Contact header on + BYE and Cancel + +2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant + + * configure.ac: fix error output when checking for openh323 to + refer to openh323 instead of pwlib (issue #8222, misaksen) + +2006-10-25 19:16 +0000 [r46252] Olle Johansson + + * channels/chan_sip.c: Somewhat ugly code to try to fix issue + #7608. Since the problem was not very well defined, the fix is a + bit fuzzy too... Thanks to Luigi for accidentally spotting the + possible problem! + +2006-10-25 19:08 +0000 [r46249] Russell Bryant + + * apps/app_queue.c: update warning message to include "agi" option + (issue #8225, jmls) + +2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming + + * sounds/Makefile: use 1.4.3 extra sounds with corrected silence + files + + * sounds/sounds.xml, sounds/Makefile: add support for prebuilt + G.722 prompts and music on hold files + +2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson + + * channels/chan_sip.c: show settings doesn't produce a list of + similar objects, it should stay a "show" + +2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming + + * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, + channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, + pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, + main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, + cdr/cdr_custom.c, channels/chan_mgcp.c, + apps/app_parkandannounce.c, apps/app_voicemail.c, + channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, + res/res_adsi.c, main/utils.c, apps/app_ices.c, + pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, + apps/app_getcpeid.c: apparently developers are still not aware + that they should be use ast_copy_string instead of strncpy... fix + up many more users, and fix some bugs in the process + +2006-10-25 04:58 +0000 [r46165] Tilghman Lesher + + * main/pbx.c: WaitExten truncates decimals of times to wait, + instead of accepting them (Bug 8208) + +2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming + + * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, + channels/chan_h323.c, channels/chan_iax2.c, + include/asterisk/frame.h: add passthrough and file format support + for G.722 16KHz audio (issue #5084, original patch by andrew, + updated by mithraen) + + * channels/chan_sip.c, main/translate.c: code zone experiment: + don't offer formats in the outbound INVITE that aren't either + passthrough or translatable + + * main/translate.c: if multiple translators are registered for the + same source/dest combination, ensure that the lowest-cost one is + always inserted earlier in the list + +2006-10-24 20:30 +0000 [r46142] Mark Spencer + + * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, + #8147) + +2006-10-24 19:29 +0000 [r46130] Joshua Colp + + * channels/chan_iax2.c: We need to initialize our scheduler pthread + condition... yes. + +2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo + + * main/http.c: merge 45152 don't leak descriptors in http.c + + * channels/chan_sip.c: merge 45966 refer_to_domain potentially + containing options + + * channels/chan_sip.c: merge 46026 improper checks on get_header() + return values + + * channels/chan_sip.c: merge 46045 prevent NULL args to + ast_strdupa() in chan_sip.c + +2006-10-24 05:23 +0000 [r46093] Russell Bryant + + * Makefile: Restore the ability to remove the firmware directory + without causing the installation to fail (issue #8111) + +2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming + + * main/translate.c: ensure that the translation matrix is properly + lock-protected every place it is used + + * include/asterisk/translate.h, main/translate.c: add an API call + to allow channel drivers to determine which media formats are + compatible (passthrough or transcode) with the format an existing + channel is already using + + * doc/imapstorage.txt: simplify and correct voicemail IMAP storage + build instructions + +2006-10-24 03:01 +0000 [r46078] Tilghman Lesher + + * main/channel.c: Pass through a frame if we don't know what it is, + rather than trying to pass a NULL, which will segfault a channel + driver (Bug 8149) + +2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant + + * utils/muted.c, utils/ael_main.c: In muted.c, check the return + value of strdup. In ael_main.c, check the return value of calloc. + (issue #8157) In passing fix a few minor bugs in ael_main.c. The + last argument to strncpy() was a hard-coded 100, where it should + have been 99. I changed this to use sizeof() - 1. + + * apps/app_meetme.c: Fix the descriptions of some of the + MeetMeAdmin options (issue #8098, mflorell) + + * res/res_jabber.c: don't crash when an incoming message has no + "from" (issue #8205, jmls) + +2006-10-23 00:27 +0000 [r45928] Joshua Colp + + * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 + lines Don't leak memory mmmk? ........ + +2006-10-22 21:44 +0000 [r45916] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 + Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and + couldn't be initialized it would cause a segfault after 'reload'. + Reported by Drew/Matt thx. ........ + +2006-10-21 18:49 +0000 [r45818] Russell Bryant + + * res/res_monitor.c: Add a couple missing unregistrations of + manager actions and remove duplicate unregistrations of + applications. (issue #8194, jmls) + +2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp + + * main/loader.c: Don't use promotion on Darwin because it doesn't + seem to work quite right in all cases, this should solve the + unresolved symbol issue people have been seeing. + + * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get + installed in the proper location (reported on asterisk-dev + mailing list) + +2006-10-20 07:44 +0000 [r45741] Olle Johansson + + * channels/chan_sip.c: Let's understand SIP: - REFER can create + dialog, Asterisk does not support it yet - NOTIFY can create + dialog in Asterisk's implementation (voicemail) even though we + don't support the server side of it. In this case, the standard + is a side issue ;-) - Added extened functionality for unsupported + methods (PING, PUBLISH) so we don't create PVT's for those + either. Russellb needs to judge what to do with this in 1.2, but + I think the current implementation n 1.2 is a bug since we're + sending bad replies to NOTIFY and REFER outside of dialogs + +2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp + + * res/res_jabber.c: Let's remember to unregister JabberStatus too + (issue #8184 reported by jmls) + + * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct + 2006) | 2 lines Respect language selection when seeing if the + file exists (issue #8178 reported by mnicholson) ........ + + * channels/chan_sip.c: If the jitterbuffer is forced on then we + can't partially bridge (reported by wangster on #asterisk-dev) + +2006-10-19 00:59 +0000 [r45622] Russell Bryant + + * channels/chan_sip.c: Don't leak the actual thread-specific + sip_pvt struct + +2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming + + * channels/chan_sip.c: don't leak memory when a chan_sip thread is + destroyed that has a thread-local temp_pvt allocated + +2006-10-18 21:03 +0000 [r45595] Joshua Colp + + * main/asterisk.c: Don't modify things if we are using vfork as + this is very bad and may cause unexpected behavior (issue #7970 + reported by Nick Gavrikov) + +2006-10-18 11:54 +0000 [r45517] Olle Johansson + + * channels/chan_sip.c: remove duplicate declarations + +2006-10-18 04:09 +0000 [r45464] Luigi Rizzo + + * main/http.c: merge from trunk: move ast_variables_destroy() to a + better place in handle_uri() to avoid leaking memory on non + existing files. + +2006-10-18 03:02 +0000 [r45452] Joshua Colp + + * main/rtp.c: Don't segfault if you're using a channel driver that + doesn't turn RTCP on + +2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant + + * main/channel.c: Don't attempt to access private data members of + the pthread_mutex_t object, because this does not work on all + linux systems. Instead, just access the reentrancy field in the + ast_mutex_info struct when DEBUG_THREADS is enabled. If + DEBUG_CHANNEL_LOCKS is enabled, the developer probably has + DEBUG_THREADS on as well. (issue #8139, me) + + * configs/sip_notify.conf.sample: update entry to reboot a snom + phone (issue #7850, pnlarsson) + +2006-10-17 Kevin P. Fleming + + * Asterisk 1.4.0-beta3 released. + +2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming + + * include/asterisk/stringfields.h, main/ast_expr2.c, + main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: + optimize the 'quick response' code a bit more... no more malloc() + or memset() for each response expand stringfields API a bit to + allow reusing the stringfield pool on a structure when needed, + and remove some unnecessary code when the structure was being + freed + +2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp + + * channels/chan_sip.c: Don't create a "real" pvt structure for + requests that shouldn't be able to create one. Instead use a + temporary pvt and fill it with enough information so we can send + a reply. + +2006-10-17 17:39 +0000 [r45329] Olle Johansson + + * configs/sip.conf.sample: Adding information about Marks + direct-RTP hack to the docs... + +2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming + + * LICENSE: provide licensing language for IAXy firmware file + +2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp + + * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new + directed pickup (BE-85). + +2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson + + * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for + your support! + + * channels/chan_sip.c: Don't destroy dialog for unexpected REFER + response... + +2006-10-14 04:38 +0000 [r45143] Steve Murphy + + * funcs/func_rand.c: update the doc string for both AEL and + extensions.conf users. + +2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming + + * main/acl.c don't drop the entire permit/deny list when an attempt + is made to add an invalid entry (BE-92) + +2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp + + * res/res_speech.c: Clear the quiet flag too since we are + restarting a recognition again (reported on -dev by Stephan + Edelman) + + * res/res_speech.c: Check return value from engine in case of + failure (ie: out of licenses) (reported on -dev mailing list) + +2006-10-13 20:52 +0000 [r45103] Steve Murphy + + * pbx/ael/ael-test/ref.ael-vtest17 (added), + pbx/ael/ael-test/ael-vtest17/extensions.ael (added), + pbx/ael/ael-test/ael-vtest17 (added), + pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in + this release via these changes + +2006-10-13 19:19 +0000 [r45088] Christian Richter + + * channels/chan_misdn.c: avoiding warning, fixing potential bug + +2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp + + * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, + codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, + codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, + codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, + codecs/lpc10/synths.c, codecs/lpc10/preemp.c, + codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, + codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, + codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, + codecs/lpc10/lpcini.c, codecs/lpc10/random.c, + codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, + codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, + codecs/lpc10/analys.c, codecs/lpc10/onset.c, + codecs/lpc10/energy.c, codecs/lpc10/deemp.c, + codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, + codecs/lpc10/median.c, codecs/lpc10/encode.c, + codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, + codecs/lpc10/invert.c: And file said... let the compiler warnings + STOP! + + * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 + reported by mnicholson) + + * apps/app_playback.c: Move say.conf existence check to do_say + function since it is called from multiple places (issue #8144 + reported by kshumard) + +2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming + + * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if + we have multiple bindings (reported on asterisk-dev) + +2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp + + * channels/chan_sip.c: Complete merging in RPID screen changes + (issue #8101 reported by hristo, patch by oej in revision 44757) + + * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add + the background refresh item back into the scheduler if enabled + since it is deleted during reload. (issue #8142 reported by + p_lindheimer) + +2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/utils.c: use a configure script test for PMTU discovery + control instead of just assuming it's available on Linux + +2006-10-13 14:45 +0000 [r44994-45026] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some + echocandisable issues when bridged. this caused a kernel panic + sometimes.. also some minor formatting fixes + + * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause + got a wrong isdn cause at RELEASE_COMPLETE + +2006-10-12 22:07 +0000 [r44992] Luigi Rizzo + + * channels/chan_sip.c: merge formatting and minor code + simplifications from trunk + +2006-10-12 20:34 +0000 [r44982] Matt O'Gorman + + * channels/chan_gtalk.c: fix for bug 7764. + +2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming + + * channels/chan_sip.c: we can only send one 'a=ptime' attribute per + media session, not one for each format + + * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, + main/utils.c: ensure that IAX2 and SIP sockets allow UDP + fragmentation when running on Linux (thanks to Brian Candler on + the asterisk-dev list for the tip) + +2006-10-12 16:56 +0000 [r44945] Russell Bryant + + * main/manager.c: fix a silly typo in a comment that I saw while + reading the commit list + +2006-10-12 16:08 +0000 [r44942] Joshua Colp + + * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue + #8135 reported by ssokol) + +2006-10-12 12:55 +0000 [r44921] Nadi Sarrar + + * main/manager.c: append_event must be called while holding the + session lock + +2006-10-12 10:24 +0000 [r44911] Russell Bryant + + * res/res_jabber.c: change some debug output to use LOG_DEBUG + instead of verbose output + +2006-10-11 16:57 +0000 [r44888] Jason Parker + + * main/db1-ast/Makefile: These are already set by the parent + Makefile.. There is no need to have this here (it doesn't + actually work anyways). + +2006-10-11 09:18 +0000 [r44854] Christian Richter + + * channels/misdn/isdn_lib.c: removed warning because of missing + prototype declaration + +2006-10-10 19:23 +0000 [r44830] Olle Johansson + + * channels/chan_sip.c: Do not set default/global values in the + variable declaration, set it in reload_config() + +2006-10-10 17:21 +0000 [r44819] Joshua Colp + + * channels/chan_sip.c: Move some stuff around so that a NOTIFY + dialog won't hang around until the end of the world under certain + circumstances + +2006-10-10 16:44 +0000 [r44809] Paul Cadach + + * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: + CHANNEL() function sometime mix parameter and value + +2006-10-10 16:42 +0000 [r44808] Tilghman Lesher + + * funcs/func_logic.c: Lost of a bit of logic when this was + simplified between 1.2 and 1.4 (Bug 8117) + +2006-10-10 16:30 +0000 [r44806] Joshua Colp + + * channels/chan_sip.c: Bail out if we have no refer structure and + we get a refer response + +2006-10-10 16:21 +0000 [r44805] Luigi Rizzo + + * channels/chan_sip.c: more merge from trunk (comments and change a + static function name) + +2006-10-10 15:23 +0000 [r44788] Joshua Colp + + * channels/chan_sip.c: Only set DTMF information if an RTP + structure exists + +2006-10-10 13:50 +0000 [r44786] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added + support of dynamically enabling hdlc on bchannels + +2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo + + * channels/chan_sip.c: whitespace changes related to previous + commit + + * channels/chan_sip.c: merge a few code simplifications that have + gone into trunk during last week, to reduce differences between + the two branches and make porting fixes easier. + +2006-10-09 16:12 +0000 [r44764] Jason Parker + + * channels/chan_skinny.c: Fix a problem where phones that go + "missing" never got unregistered. Issue #8067, reported by pj, + patch by Anthony LaMantia (with minor whitespace modifications) + +2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp + + * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid + the deadlock + + * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup + (issue #8115 reported by vazir) + +2006-10-08 14:14 +0000 [r44746] Luigi Rizzo + + * channels/chan_sip.c: do not dereference p if we + know it is NULL + +2006-10-07 14:39 +0000 [r44684] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate + caller's transfer capability too + +2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo + + * channels/chan_sip.c: put common code in a + function to avoid repetitions. + + * channels/chan_sip.c: remove hardwired usage of 5060, use + DEFAULT_SIP_PORT instead + + * channels/chan_sip.c: option_debug checking + before printing to debug channel. + + * channels/chan_sip.c: backport simplifications on sip_register, + usage of ast_set2_flag(), and fixes to the handling of failed + module loading. + + * channels/chan_sip.c: improve and document function + get_in_brackets(), introducing a helper function + find_closing_quote() of more general use. + +2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming + + * include/asterisk/linkedlists.h: ensure that mutex locks inside + list heads are initialized properly on platforms that require + constructor initialization (issue #8029, patch from timrobbins) + + * CHANGES: remove Jingle as per mog + +2006-10-06 21:08 +0000 [r44628] Joshua Colp + + * main/rtp.c: Remove the seqno check for RFC2833, the handler is + smart enough to not need it. + +2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming + + * CHANGES: various cleanups + +2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp + + * main/rtp.c: When the sequence number rolls over then reset the + recorded sequence number for DTMF (issue #8106 reported by + bungalow) + + * main/file.c: Even more frames to treat as though the remote side + disappeared (issue #8097 reported by eldadran) + +2006-10-06 15:59 +0000 [r44567] Luigi Rizzo + + * main/manager.c, main/http.c: make sure sockets are blocking when + they should be blocking. + +2006-10-06 12:53 +0000 [r44559-44563] Christian Richter + + * channels/chan_misdn.c: fixed segfault which happens during + hold/transfer action + + * channels/chan_misdn.c: if INFORMATION Message come with keypad + instead of called party number, we just use the keypad as called + party number. + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + added the option 'reject_cause' to make it possible to set + the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, + which is automatically rejected because chan_misdn does not + support that kind of callwaiting. Therefore chan_misdn supports + now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc + now gets the info if the requested channel is incoming or + outgoing to make the 3. channel possible + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, + removed a useless bc field, added setting of frame.delivery fields, + some minor code cleanups + +2006-10-05 19:57 +0000 [r44502] Joshua Colp + + * main/file.c: Treat busy control frames as hangup in the file streaming + core (issue #8097 reported by eldadran) + +2006-10-05 18:21 +0000 [r44488] Steve Murphy + + * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. + Many thanks to Doug! + +2006-10-05 18:01 +0000 [r44486] Joshua Colp + + * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite + hanging by a thread if the other side is already setup with T.38 + +2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming + + * main/app.c: don't segfault when an argument without a close + parenthesis is found stop parsing as soon as that situation + occurs + +2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy + + * CHANGES: I put the accumulated changes from the commit logs and + inspection, into CHANGES. Hope everyone approves! + + * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the + install process sticks muted.conf in /etc/asterisk, so that's + where muted should look for it, right? + +2006-10-05 02:40 +0000 [r44450] Joshua Colp + + * channels/chan_sip.c: Don't totally bail out if T.38 was + negotiated + +2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming + + * channels/chan_sip.c: fix Polycom presence notification again + +2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo + + * utils/Makefile: as far as i can tell astman only uses newt... + + * Makefile: put linker flags in ASTLDFLAGS where they belong + +2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming + + * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE + requests add workaround for new Polycom firmware SUBSCRIBE + requests (bug is known to exist in 2.0.1 firmware) + + * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually + work + +2006-10-04 19:57 +0000 [r44380] Steve Murphy + + * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, + pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, + pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, + pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, + pbx/ael/ael-test/ael-test16/extensions.ael (added), + pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, + pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, + pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, + pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the + problems reported in bug 8090 + +2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming + + * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, + main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, + main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, + include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, + main/devicestate.c, main/utils.c, res/res_musiconhold.c, + channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update + thread creation code a bit reduce standard thread stack size + slightly to allow the pthreads library to allocate the stack+data + and not overflow a power-of-2 allocation in the kernel and waste + memory/address space add a new stack size for 'background' + threads (those that don't handle PBX calls) when LOW_MEMORY is + defined + +2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy + + * configs/muted.conf.sample: I've been meaning to add some + explanation about muted... here it is + + * configs/manager.conf.sample: CLI reverbification update to this + config file + + * apps/app_macro.c: In response to bug 7776, a Warning has been + added to the doc string for Macro(). + +2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming + + * main/asterisk.c, main/loader.c, main/term.c, Makefile, + include/asterisk.h: ensure that local include files are always + used avoid a duplicate function name (term_init()) + +2006-10-03 22:35 +0000 [r44312] Matt O'Gorman + + * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing + client without resource. + +2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming + + * apps/app_queue.c: fix a logic error in my previous fix to the queue + reload code + +2006-10-03 18:42 +0000 [r44286] Paul Cadach + + * channels/h323/ast_h323.cxx: Change default presentation indicator + to "user provided not screened" if octet 3a missed in + CallingPartyNumber IE + +2006-10-03 18:35 +0000 [r44284] Joshua Colp + + * channels/chan_sip.c: Use VideoSupport instead so it is considered + a valid XML attribute name. (issue #8075 reported by renemendoza) + +2006-10-03 18:30 +0000 [r44283] Paul Cadach + + * channels/h323/ast_h323.cxx: Fix preparation of type and + presentation of calling number + +2006-10-03 00:01 +0000 [r44240] Matt O'Gorman + + * doc/jingle.txt, channels/chan_jingle.c (removed), + include/asterisk/jabber.h, configs/jingle.conf.sample (removed), + res/res_jabber.c: updated res_jabber for even better component + support, soon will be jep-0100 compliant. also removed + chan_jingle and infromed info from jingle.txt, chan_gtalk still + works and should be used in this version. + +2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp + + * channels/chan_sip.c: Change the fd on the I/O context in case it + changed during the reload, which is indeed possible. (issue #7943 + reported by eclubb) + + * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN + instead of hardcoding the path for the error message (issue #7942 + reported by eclubb) + +2006-10-02 18:52 +0000 [r44186] Paul Cadach + + * configs/users.conf.sample, pbx/pbx_config.c: Missed part of + userconf functionality for chan_h323 + +2006-10-02 17:25 +0000 [r44169] Joshua Colp + + * main/io.c: Shrink when current_ioc is unused. It is set to -1 when + unused, not 0. (issue #7941 reported by eclubb) + +2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach + + * doc/realtime.txt: Typo fix + + * channels/chan_h323.c: Optimization of oh323_indicate(): less + locks - less problems, plus single exit point + +2006-10-02 02:38 +0000 [r44146] Mark Spencer + + * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when + you're not talking about a channel :) + +2006-10-01 19:32 +0000 [r44135] Paul Cadach + + * channels/chan_h323.c: Do not simulate any audio tones if we got + PROGRESS message + +2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant + + * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to + be empty. The cause is that since ASTDATADIR is explicitly + exported using "export ASTDATADIR" at the top of the Makefile, + make no longer considers the variable "undefined", so the + Makefile can't use ?= to set ASTDATADIR if not yet set. (issue + #8063, reported by akohlsmith, fixed by me) + + * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" + option in the sample queues.conf (issue #8065, adamg) + +2006-10-01 15:01 +0000 [r44109] Luigi Rizzo + + * channels/chan_sip.c: sync with trunk - move variable declarations + to the beginning of a block. + +2006-09-30 19:20 +0000 [r44090] Paul Cadach + + * main/rtp.c: Allow one-way RTP streams (device->Asterisk) + +2006-09-30 16:28 +0000 [r44080] Luigi Rizzo + + * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent + build problems: - with AST_DEVMODE, building codecs/lpc10 fails + because of lots of warnings, and the configure step in editline + fails as well. Fix this by removing the -Werror in these steps. - + on FreeBSD (but probably on other platforms as well), the final + link of asterisk fails because AST_LIBS was not exported to the + subdirs Makefiles. Add a proper fix in the top-level Makefile (a + possible alternative way is to add "export AST_LIBS" near the + beginning of the file). With this fix, i believe that some of the + platform-specific conditionals in main/Makefile are redundant + (because they should be already dealt with in the top level + Makefile) but i don't have a platform to check. Merging to head + will happen in a moment. + +2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach + + * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment + of previous fix: Issue #7928 - Don't send both 404 and 503. Fix + by phsultan with a small fix by me, myself or I. Thanks, + Philippe! (This was caused by my changes to the transaction + handling) + + * channels/chan_sip.c: Found some buggy SIP clients (phones Planet + VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which + sends ACK not on OK message only (when remote party answers) but + on RINGING message too, so when we send 200 OK message, we get + unidentified ACK message (because INVITE acknowledged on RINGING + message already), so 200 OK retransmits within its retransmission + interval then call gets dropped. If someone else knows how to + provide workaround for such cases, please, fix it in correct way. + Thanks to ssh from #asteriskru for provide access to his box to + study and fix this case. + +2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming + + * agi, utils: ignore temporary files made by the Makefiles during a + build + + * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, + codecs/Makefile, utils/Makefile, configure, + build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, + Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, + pbx/Makefile, res/Makefile, channels/Makefile: fix a few build + system bugs, and convert Makefiles to be compatible with GNU make + 3.80 + +2006-09-29 22:35 +0000 [r44053] Jason Parker + + * main/asterisk.c, main/cli.c: Fix a bug with the removal of + 'atleast' argument to 'core verbose' and 'core debug'. Add that + argument back in. + +2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach + + * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more + carefully when no CallingNumber IE available + + * channels/h323/ast_h323.cxx: Fake display name by called number on + incoming calls (until passing connected number/connected name is + not implemented) + + * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add + includes + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly + pass TON/PRESENTATION information - original + H323Connection::SendSignalSetup() destroys Q.931 fields. + +2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming + + * main/Makefile: yet another place where we were not using the + correct CFLAGS by default + + * main/Makefile: missed one conversion to ASTCFLAGS + +2006-09-29 18:30 +0000 [r44009] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass + TON/PRESENTATION information too + +2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming + + * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, + main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, + Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse + CFLAGS and LDFLAGS for build of Asterisk components, because they + are also then used for non-Asterisk components (like menuselect); + use our own variables instead + + * configure, configure.ac: support --without-curl in configure + script + + * Makefile.rules: another cross-compile fix + + * Makefile: a couple more environment settings that can't leak into + the menuselect build + + * main/cli.c: proper fix for ast_group_t change + + * include/asterisk/lock.h: eliminate compiler warning when + DEBUG_CHANNEL_LOCKS is enabled and users of this header file + don't also include channel.h + +2006-09-28 20:11 +0000 [r43944] Jason Parker + + * apps/app_queue.c: Fix incorrect argument order for member names, + on persisted members. Issue 8047, patch by jmls. + +2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp + + * apps/app_playback.c, res/res_monitor.c, + include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, + channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/udptl.c, main/frame.c, funcs/func_timeout.c, + channels/chan_sip.c, apps/app_festival.c, + channels/iax2-provision.c, apps/app_alarmreceiver.c, + res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: + Put in missing \ns on the end of ast_logs (issue #7936 reported + by wojtekka) + +2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming + + * apps/app_queue.c: fix buggy (and overly complex) loop used during reload + of app_queue for static member list updating + +2006-09-28 17:34 +0000 [r43918] Paul Cadach + + * channels/h323/ast_h323.cxx: Extend call establishment timeout + +2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp + + * channels/chan_iax2.c: Make sure the pvt exists before accessing + it again as it may have gone away (issue #7562 reported by Seb7 + and issue #7939 reported by sorg) + + * main/cli.c: Warning be gone! + +2006-09-28 16:41 +0000 [r43899] BJ Weschke + + * apps/app_queue.c: app_queue is comparing the device names incorrectly + while checking their statuses. It's internal list of interfaces + includes the dial string, while the argument passed to this + function does not have the dial string (/n for a local channel). + This causes it to ignore the device state changes because it + thinks it belongs to none of its members. (#8040 reported and + patch by tim_ringenbach) + +2006-09-28 16:17 +0000 [r43893] Joshua Colp + + * apps/app_meetme.c: Stop the stream after waitstream returns so that our + formats get restored. (issue #7370 reported by kryptolus) + +2006-09-28 15:56 +0000 [r43877] Paul Cadach + + * channels/h323/ast_h323.cxx: Fix compiler warning + +2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke + + * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - + tim_ringenbach reported and patched) + + * apps/app_queue.c: Autopause not working for queue members. (#8042 + - jmls reported and patch) + +2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force + remote side to start media on outgoing PROGRESS message + + * include/asterisk/compiler.h: Put attribute tag at correct place + +2006-09-28 11:03 +0000 [r43852] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed a bug which led to chan_list zombies, + when the call could not be properly established in misdn_call. + also removed the ACK_HDLC stuff which is not really needed. + +2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach + + * channels/h323/ast_h323.cxx: Do not open transmit channel until + TCS is received + + * main/file.c: Don't warn on HOLD/UNHOLD control frames + + * main/file.c: Don't treat unknown control frames as voice + +2006-09-27 20:21 +0000 [r43816] Tilghman Lesher + + * apps/app_voicemail.c: Avoid inability to lock directory log message by + creating the directory ahead of time. (Issue 7631) + +2006-09-27 19:44 +0000 [r43801-43803] Jason Parker + + * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS + not being set under certain circumstances. Fix a minor issue, to + make it use the filenames that were parsed, instead of the entire + argument string. Fix Background() to return -1 like Playback(), + if no args are specified. + +2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp + + * main/rtp.c: Compensate for out of order packets better if RFC2833 + compensation is turned on. + + * channels/chan_iax2.c: Get rid of two functions from a time now + past (we THINK these are from pre-recursive lock time) that may + be contributing to two open issues on the bug tracker (7562/7939) + and that has the potential to just make bad things happen if the + timing is right. + +2006-09-27 16:55 +0000 [r43779] Russell Bryant + + * main/channel.c,res/res_features.c: Fix a problem that occurred if + a user entered a digit + that matched a bridge feature that was configured using multiple + digits, and the digit that was pressed timed out in the feature + digit timeout period. For example, if blind transfer is + configured as '##', and a user presses just '#'. In this + situation, the call would lock up and no longer pass any frames. + (issue #7977 reported by festr, and issue #7982 reported by + michaels and valuable input provided by mneuhauser and kuj. Fixed + by me, with testing help and peer review from Joshua Colp). There + are a couple of issues involved in this fix: 1) When + ast_generic_bridge determines that there has been a timeout, it + returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets + this result, it calls ast_generic_bridge over again with the same + timestamp for the next event. This results in an endless loop of + nothing until the call is terminated. This is resolved by simply + changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it + sees a timeout. 2) I also changed ast_channel_bridge such that if + in the process of calculating the time until the next event, it + knows a timeout has already occured, to immediately return + AST_BRIDGE_COMPLETE instead of attempting to bridge the channels + anyway. 3) In the process of testing the previous two changes, I + ran into a problem in res_features where ast_channel_bridge would + return because it determined that there was a timeout. However, + ast_bridge_call in res_features would then determine by its own + calculation that there was still 1 ms before the timeout really + occurs. It would then proceed, and since the bridge broke out and + did *not* return a frame, it interpreted this as the call was + over and hung up the channels. The reason for this was because + ast_bridge_call in res_features and ast_channel_bridge in + channel.c were using different times for their calculations. + channel.c uses the start_time on the bridge config, which is the + time that the feature digit was recieved. However, res_features + had another time, 'start', which was set right before calling + ast_channel_bridge. 'start' will always be slightly after + start_time in the bridge config, and sometimes enough to round up + to one ms. This is fixed by making ast_bridge_call use the same + time as ast_channel_bridge for the timeout calculation. ........ + +2006-09-27 16:24 +0000 [r43775] Christian Richter + + * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn + versioning, since Asterisk has it's own + +2006-09-27 16:23 +0000 [r43774] Joshua Colp + + * channels/chan_sip.c: Make rfc2833compensate a global option. + +2006-09-27 04:35 +0000 [r43756] Russell Bryant + + * apps/app_voicemail.c: Backport revision 43754 from the trunk, + which removes an unused buffer from mm_login to close bug 8038, + as well as addresses some formatting and coding guidelines issues + in passing. Originally, I did not commit this to 1.4 since it is + not necessarily fixing a bug. However, since the IMAP storage + code is brand new, I decided it would be better to make the + change here as well, in case someone has to work on this code to + address issues in the very near future. I don't want to make + unnecessary merge problems going to the trunk. + +2006-09-27 02:32 +0000 [r43739] Steve Murphy + + * configs/extensions.ael.sample: This change to extensions.ael was + to fix bug 8031; the install scripts are causing it to be copied + to /etc/asterisk/extensions.ael, and because it is a fairly + direct conversion of the original extensions.conf, the macro and + context names clash with the existing extensions.conf. So, I put + an ael- in front of all macros and contexts, and checked every + goto and macro call. Also, this file compiles under aelparse. + +2006-09-26 20:56 +0000 [r43710] Russell Bryant + + * main/asterisk.c: Back in revision 4798, this message was changed from + using ast_cli() to directly calling write(). During this change, + checking if this was a remote console was removed. This caused + this message about using "exit" or "quit" to exit an Asterisk + console to come up in times where it did not make sense. This + change restores the check to see if this is a remote console + before printing the message. (fixes BE-65) + +2006-09-26 20:47 +0000 [r43707] Joshua Colp + + * .cleancount, main/cli.c, channels/chan_sip.c, + include/asterisk/channel.h: Use proper type to represent the group variable + (issue #8025 reported by makoto) + +2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant + + * channels/chan_sip.c: Add missing newline character in the warning + message about deprecated TOS values in configuration. + + * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain + mailbox definitions, don't introduce a length limit on the + definition by using a 256 byte temporary storage buffer. Instead, + make the temporary buffer just as big as it needs to be to hold + the entire mailbox definition. (fixes BE-68) + +2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp + + * channels/chan_local.c: Strip options off the argument passed for + devicestate in chan_local. (issue #8034 reported by pcardozo) + + * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight + overhaul of the whisper support. 1. We need to duplicate the + frame from ast_translate 2. We need to ensure we always have + signed linear coming in for signed linear combining. 3. We need + to ensure we are always feeding signed linear out. 4. Properly + store and restore write format when beeping on the channel we are + whispering on. 5. Properly discontinue the stream on the channel + for the beep. (issue #8019 reported by timkelly1980) + +2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming + + * sounds/Makefile: update to use 1.4.3 core sounds, with corrected + beep/beeperr/tt-monkeys files + +2006-09-26 18:08 +0000 [r43650-43674] Jason Parker + + * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by + Dan Austin. Maximum values were incorrect, which is why this is + being put in 1.4 + + * channels/chan_skinny.c: Add proper codec support to chan_skinny. + Works with at least ulaw, alaw, and g729a. This is technically a + "new feature", but there are justifications for it. I found a bug + with the recent rtp packetization changes, which caused the media + setup to fail under certain circumstances, particularly when + using allow=all, or having no allow= statements (globally or on + the device). I could have either removed the rtp packetization + features, or I could add proper codec support (which, without, I + think most people would consider to be a bug anyways). + +2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher + + * apps/app_voicemail.c: Should have moved these lines up in the + merge, instead of removing them + + * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) + delete=yes was ignored 2) maxmessages was ignored + +2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach + + * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, + channels/h323/cisco-h225.asn: Fix ASN1 description of + non-standard Cisco extensions + + * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport + changes of trunk: 1) r43540: Avoid possible deadlock on channel + destruction 2) r43590: Disable fastStart if requested by remote + side + +2006-09-25 15:23 +0000 [r43616] Jason Parker + + * sounds/Makefile: One more fix for sounds installation - this time + for portability. Reported to asterisk-dev mailing list. + +2006-09-25 14:52 +0000 [r43605] Steve Murphy + + * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from + crashing if trying to play an OGG moh file. + +2006-09-25 06:15 +0000 [r43582] Paul Cadach + + * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, + channels/chan_h323.c: Merged revisions 43472,43495 from trunk + +2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant + + * channels/iax2-provision.c: Fix a CLI command registration issue + where an erroneous message claiming that "iax2 show provisioning" + was already registered. This was because this command was + registering itself as both the command, as well as the command it + is deprecating. (issue #8022, reported by bjweeks, fixed by + myself) + + * channels/chan_iax2.c:Check to see if the channel that is activating the + IAXPEER function is actually an IAX2 channel before proceeding to + process it to avoid crashing. (issue #8017, reported by admott, + fixed by myself) + +2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming + + * Makefile: don't output the 'build complete' message when the + target being run is already going to do an installation + +2006-09-22 22:12 +0000 [r43518] Jason Parker + + * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded + properly. Remove reload support, since it doesn't + actually...work. + +2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy + + * pbx/pbx_ael.c: This commits a change to return + MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all + goes well for bug 8004 + + * pbx/pbx_ael.c: If the extensions.ael file not found, or + unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. + +2006-09-22 17:25 +0000 [r43492] Jason Parker + + * main/cli.c: Make sure we explicitly set the CLI command to not be + deprecated, if it isn't. + +2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming + + * sounds/Makefile: use rebuilt extra sounds + + * main/channel.c: all the Linux systems I have don't use + '__m_count' for this field, so I don't know where this came + from... + +2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant + + * include/asterisk/threadstorage.h: backport the compatability fix + to use attribute_malloc instaed of __attribute__ ((malloc)) + + * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN + could not be configured (issue #8006, Mithraen) + + * main/frame.c: Suppress a compiler warning about the use of a + potentially uninitialized variable. It couldn't actually happen, + though. + +2006-09-22 03:01 +0000 [r43469] Jason Parker + + * channels/chan_skinny.c: First shot at unload_module in + chan_skinny.. More to come. + +2006-09-21 23:50 +0000 [r43466] Matt O'Gorman + + * include/asterisk/jabber.h, channels/chan_gtalk.c, + res/res_jabber.c: updates for better compontent support + +2006-09-21 23:24 +0000 [r43464] Tilghman Lesher + + * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we + actually documented how the new features in res_odbc actually + work. (Oops) + +2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp + + * channels/chan_oss.c: Some more clean up in the load function for + chan_oss (issue #8002 reported by Mithraen with minor mods by + moi) + + * channels/chan_mgcp.c: Clean up chan_mgcp's module load function + (issue #8001 reported by Mithraen with mods by moi) + +2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming + + * main/Makefile, build_tools/strip_nonapi (added): add another + attempt to strip non-API symbols from the final binary... script + will need to be extended to work on non-Linux systems + +2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher + + * apps/app_url.c: Fix documentation to reflect how Url() really + works + + * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates + +2006-09-21 Kevin P. Fleming + + * Asterisk 1.4.0-beta2 released. + +2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming + + * main/Makefile: remove this change... it requires binutils 2.17 + +2006-09-20 23:19 +0000 [r43396] Jason Parker + + * build_tools/make_version: fix minor typo in the way version is + handled + +2006-09-20 Kevin P. Fleming + + * Asterisk 1.4.0-beta1 released. -- cgit v1.2.3