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2010-06-22Merged revisions 271903 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271904 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271690 via svnmerge from mnicholson1-125/+124
https://origsvn.digium.com/svn/asterisk/trunk ................ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271691 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Merged revisions 269307 via svnmerge from jpeeler1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010) | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271338 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270981 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r270981 | qwell | 2010-06-16 16:10:48 -0500 (Wed, 16 Jun 2010) | 11 lines Merged revisions 270980 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270982 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270867 via svnmerge from dvossel1-6/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r270867 | dvossel | 2010-06-16 12:36:51 -0500 (Wed, 16 Jun 2010) | 28 lines Merged revisions 270866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines fixes chan_iax2 race condition There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270868 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Merged revisions 270658 via svnmerge from twilson1-21/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r270658 | twilson | 2010-06-15 15:18:04 -0500 (Tue, 15 Jun 2010) | 20 lines Make contactdeny apply to src ip when nat=yes chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 Review: [full review board URL with trailing slash] ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270693 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269497 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r269497 | russell | 2010-06-09 17:19:20 -0500 (Wed, 09 Jun 2010) | 9 lines Merged revisions 269495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269502 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Merged revisions 268817 via svnmerge from tilghman1-4/+38
https://origsvn.digium.com/svn/asterisk/trunk ........ r268817 | tilghman | 2010-06-07 17:47:13 -0500 (Mon, 07 Jun 2010) | 9 lines Mailbox list would previously grow at each reload, containing duplicates. Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268819 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Fix some chan_oss build system problems (these changes were already in trunk).russell1-0/+1
This is what caused a bunch of tests to fail on 1.6.2. They expected a console channel driver, but chan_oss was failing to load. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268815 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-06Merged revisions 268495 via svnmerge from tilghman1-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r268495 | tilghman | 2010-06-05 19:37:30 -0500 (Sat, 05 Jun 2010) | 2 lines Finally track down and eliminate the "FRACK! warnings from chan_iax2". ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267537 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r267537 | russell | 2010-06-03 12:31:41 -0500 (Thu, 03 Jun 2010) | 2 lines Don't stop Asterisk if chan_usbradio isn't configured. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267490 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r267490 | russell | 2010-06-03 12:05:30 -0500 (Thu, 03 Jun 2010) | 2 lines Remove a line that was killing Asterisk on startup. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267491 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267445 via svnmerge from russell1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r267445 | russell | 2010-06-03 09:48:09 -0500 (Thu, 03 Jun 2010) | 2 lines Comment out a rule that likes to run implicitly unnecessarily, breaking builds ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267446 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267352 via svnmerge from russell2-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r267352 | russell | 2010-06-02 17:46:37 -0500 (Wed, 02 Jun 2010) | 7 lines try to fix some random chan_h323 compilation failures After some debugging, the random chan_h323 build failures appear to be due to complications introduced by some chan_h323 specific build stuff getting triggered during a clean. Simplify this by moving the h323 clean commands down into channels/makefile. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267353 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Merged revisions 266292 via svnmerge from dvossel1-11/+25
https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines fixes crash when creation of UDPTL fails (closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266293 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Merged revisions 266006 via svnmerge from dvossel1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines fixes failed SIP Directed pickup resulting in dead channel (closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Recorded merge of revisions 265842 via svnmerge from mmichelson1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, 26 May 2010) | 9 lines Re-enable "always" option for videosupport option in sip.conf. (closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265890 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265698 via svnmerge from mmichelson1-16/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 lines Properly use peer's outboundproxy for outbound REGISTERs. The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265699 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25fixes build issue with zapteldvossel1-0/+2
(closes issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265615 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265451 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, 24 May 2010) | 8 lines Print openh323 log to the Asterisk console. (closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265452 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265449 via svnmerge from mmichelson1-11/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 lines Allow type=user SIP endpoints to be loaded properly from realtime. (closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 265087 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 lines Be sure to set the sin_family on the proxy when allocating. (closes issue #17157) Reported by: stuarth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265088 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264452 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264331 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines fixes crash in check_rtp_timeout During deadlock avoidance the sip dialog pvt is locked and unlocked. When this occurs we have no guarantee the pvt's owner is still valid. We were trying to access the pvt's owner after this without checking to see if it still existed first. (closes issue #17271) Reported by: under Patches: check_rtp_timeout.diff uploaded by under (license 914) Tested by: dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264332 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17backport of DAHDI dynamic buffer policy dialstring optiondvossel1-44/+75
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@263293 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-13Merged revisions 263069 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010) | 1 line Fix inverted logic in cli command: ss7 set debug on/off ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@263070 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-13Merged revisions 262897 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010) | 4 lines Fix an off by one error that causes a crash. Thanks to Raymond Burke for pointing it out. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@262898 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-11Merged revisions 262414 via svnmerge from pabelanger1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 | pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 lines Improve logging information for misconfigured contexts (closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@262415 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-10Merged revisions 262236 via svnmerge from dvossel1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) | 11 lines fixes crash in chan_console There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@262237 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261560 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 | tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@261563 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261451 via svnmerge from tzafrir1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | 4 lines When failing to configure, don't destroy 'cfg' twice Fixes a crash when some config section had an incorrect channel config. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@261453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Merged revisions 261314 via svnmerge from pabelanger1-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines Merged revisions 261274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@261315 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 254450 via svnmerge from mnicholson1-19/+94
https://origsvn.digium.com/svn/asterisk/trunk ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@260884 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Merged revisions 260437 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@260441 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Merged revisions 260231 via svnmerge from rmudgett1-12/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@260234 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259957 via svnmerge from mmichelson1-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@259959 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259870 via svnmerge from dvossel1-5/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@259899 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259538 via svnmerge from rmudgett1-4/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@259616 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259307 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@259310 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Merged revisions 258934 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 | lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines Small error in the T.140 RTP port verbose log. (closes issue #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) Tested by: russell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@258935 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Merged revisions 258305 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@258314 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257493 via svnmerge from tilghman1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@257510 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Merged revisions 257191 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@257210 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06CallerID channel DAHDI port FXS are empty after the first call.rmudgett1-7/+21
The bug is exposed if MFC/R2 support is built into asterisk (i.e., openr2.h is present in the include path). Code that unconditionally clears the CallerID name and number is included. Also fixed a malformed if test in mkintf() added by issue 15883. Converted the if statement to a switch statement for clarity. Regression of the issue 15883 fix. (closes issue #16968) Reported by: grecco Patches: issue16968.patch uploaded by rmudgett (license 664) (closes issue #16747) Reported by: viniciusfontes git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@256368 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06Merged revisions 256265 via svnmerge from rmudgett1-7/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@256268 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Merged revisions 256015 via svnmerge from russell1-24/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@256018 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-30Merged revisions 255410 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@255413 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Merged revisions 254718 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines chan_usbradio depends on alsa. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@254721 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Initialize stream to avoid a compilation error.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@254548 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix potential crashes from trying to reference nonexistent RTP streams.mmichelson1-7/+14
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@254542 f38db490-d61c-443f-a65b-d21fe96a405b
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