1718 Commits (7351db2bd1505408e900103fa928fdc2902ea078)

Author	SHA1	Message	Date
Kevin P. Fleming	f43a65fd21	Merged revisions 208548 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	791d4f0478	Merged revisions 208464 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b2e3c3e436	Merged revisions 205696 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d1fb490d7a	Merged revisions 205600 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f22cf5c484	Merged revisions 205479 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	308c62cf59	Merged revisions 205412 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dedd3645ab	Merged revisions 205216 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d38c8395b4	Merged revisions 205120 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	6547190dd4	SIP Dialog ref counting ... This patch adds reference counting for sip dialogs into 1.6.0. When proc_session_timer() is called from the scheduler thread it has no guarantee the session timer's dialog won't be freed from underneath it. Now the session timer holds a reference to the dialog, preventing it from being destroyed during the middle of proc_session_timer(). (closes issue #13623) Reported by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d7306096a8	Merged revisions 204710 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fc33f7b57e	Merged revisions 203699 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	af930c11b3	Merged revisions 201262 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	968108c25c	Merged revisions 201056,201090 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines 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/ ........ ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	95157288a2	Merged revisions 200764 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1e91c9f3bc	Merged revisions 199857 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	046332e03e	Merged revisions 199630 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines 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 ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	8340a34a4a	Merged revisions 199051 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines 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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9c6652d306	Merged revisions 198856 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines 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 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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	95fac13256	Merged revisions 198072 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a66b938920	Merged revisions 197606 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c7731d3489	Merged revisions 197543 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	76b9a6b5d8	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	e5d7298863	Merged revisions 195882 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4892be94a6	Recorded merge of revisions 195370 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	3f94c3bd6b	Merged revisions 192357 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	10a2c0099e	Merged revisions 192318 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	60280df797	Merged revisions 192279 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5e04e9d398	Merged revisions 191367 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	53843dfb82	Merged revisions 190725 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines 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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08c9634115	Merge of timedwrlock detection broke build. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	94b71d104f	Merged revisions 190093 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	62f5546319	Merged revisions 190057 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	a0c3c3dad9	Merged revisions 189629 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r189629 | dbailey | 2009-04-21 09:28:04 -0500 (Tue, 21 Apr 2009) | 10 lines Merged revisions 189601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@189630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3d7bac20f6	typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9a69733bc2	Merged revisions 187483 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ec49f90f69	Merged revisions 187302 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	01dc8d8ea9	Merged revisions 186321 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ce37a27180	Merged revisions 184147 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	90b97eddc4	Merged revisions 183436 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	70eaf71dfe	Merged revisions 183242 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9628d62ffa	Merged revisions 183172 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e047ec4d72	Merged revisions 182847 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	c0219aa890	Merged revisions 182525 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	368b57494b	Merged revisions 181135 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2a877c8fcb	Merged revisions 180373 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines 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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9cad0b7e22	Merged revisions 180032 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	aac9d07849	Merged revisions 177732 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines 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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e938398385	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 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	36f41cee56	Merged revisions 177664 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fff4e4f7cd	Merged revisions 177387 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines Fix another merge error from 176708 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago