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2009-10-22Merged revisions 225360 via svnmerge from tilghman3-6/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225307 via svnmerge from dvossel1-10/+77
https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225311 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225033 via svnmerge from dvossel4-7/+47
https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225310 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 224932 via svnmerge from russell7-115/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224933 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Merged revisions 224856 via svnmerge from tilghman1-4/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224857 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Merged revisions 224774 via svnmerge from file1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224775 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224671 via svnmerge from kpfleming1-9/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224672 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224567 via svnmerge from file1-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224568 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224448 via svnmerge from tilghman1-2/+18
https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes. This corrects an issue reported on the -users list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224449 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17fix typo, sorryjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224337 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17Merged revisions 224331 via svnmerge from jpeeler1-2/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224332 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-16Merged revisions 224261 via svnmerge from rmudgett1-4/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224262 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-15Merged revisions 224178 via svnmerge from jpeeler1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224179 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223832 via svnmerge from jpeeler1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223833 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223756 via svnmerge from dvossel1-19/+36
https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223759 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223652 via svnmerge from kpfleming2-3/+55
https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223653 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-11Merged revisions 223487 via svnmerge from russell1-5/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223488 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Fix interpretation of PRIREDIRECTIONREASON set by chan_sip.jpeeler2-2/+4
This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223404 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223330 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223331 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223273 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223276 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223215 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223226 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223206 via svnmerge from dvossel1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223210 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223136 via svnmerge from mnicholson1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223172 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223132 via svnmerge from dvossel1-12/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223135 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223088 via svnmerge from dvossel1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223091 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222880 via svnmerge from russell4-68/+38
https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222881 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222873 via svnmerge from dvossel2-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222876 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222799 via svnmerge from rmudgett1-4/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222800 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07Merged revisions 222692 via svnmerge from rmudgett1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222693 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07Properly initialize ast_devstate_aggregate so we don't crash sporadically.seanbright1-0/+2
This looks like it was just missed during a merge. (closes issue #15841) Reported by: amorsen Patches: ast_devstate_aggregate_init-in-ast_extension_state2.patch uploaded by amorsen (license 676) Tested by: amorsen (closes issue #15852) Reported by: amorsen Tested by: amorsen, farisraouf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222605 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07Merged revisions 222543 via svnmerge from dvossel1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222546 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07Small typo (thanks, jpeeler)tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222541 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222463 via svnmerge from jpeeler1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222464 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Fix potential crash when entire span request is received.jpeeler1-2/+2
The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) Modified: branches/1.4/channels/chan_dahdi.c git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222395 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222351 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222352 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222298 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222302 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Recorded merge of revisions 222273 via svnmerge from tilghman1-20/+34
https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222279 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222176 via svnmerge from kpfleming8-20/+138
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222185 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Recorded merge of revisions 222110 via svnmerge from kpfleming5-55/+131
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 222030 via svnmerge from dvossel1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222038 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 221971 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221972 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Revert XML docs that ended up in the 1.6.0 and 1.6.1 branches during a merge.seanbright1-320/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221963 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 221920 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221921 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 221844 via svnmerge from rmudgett2-41/+66
https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221853 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 221777 via svnmerge from tilghman3-20/+21
https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221778 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221697 via svnmerge from dvossel1-9/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221745 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221705 via svnmerge from tilghman1-88/+86
https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221742 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Fixes issue with non dynamic hosts not being set for peersdvossel1-12/+12
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221712 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221554,221589 via svnmerge from mnicholson1-6/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221662 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221592 via svnmerge from kpfleming3-12/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221598 f38db490-d61c-443f-a65b-d21fe96a405b
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