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2010-11-16Use autotagged externalsv1.8.1-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.1-rc1@295163 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Importing release summary for 1.8.1-rc1 release.lmadsen2-0/+929
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.1-rc1@295162 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Importing files for 1.8.1-rc1 release.lmadsen3-0/+26287
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.1-rc1@295161 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Creating tag for the release of asterisk-1.8.1-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.1-rc1@295160 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-15Merged revisions 295062 via svnmerge from tilghman1-0/+191
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines Create test verifying results of expression parser ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@295078 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-15Merged revisions 294988 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) | 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294989 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294910 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) | 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294911 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294904 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294905 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294822 via svnmerge from rmudgett2-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Remove CCSS architecture PDF.russell1-0/+0
It has been moved to: https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294745 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Remove most of the contents of the doc dir in favor of the wiki content.russell94-20765/+15
This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294740 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294733 via svnmerge from jpeeler1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294734 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294639 via svnmerge from jpeeler3-113/+192
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294640 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Fixing the Mac OS X build (bamboo warning)tilghman1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294605 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Properly queue files with inotify(7).tilghman1-59/+101
(closes issue #18089) Reported by: abelbeck Patches: 20101021__issue18089.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Tweak a couple of CLI commands back to their original form.russell3-4/+8
The "module" in this case is two parts, so there are two words before the verb of the CLI command. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294535 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Merged revisions 294500 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) | 7 lines Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294501 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Allow ast_do_masquerade() failure to be reported again.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294466 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Merged revisions 294429 via svnmerge from tilghman3-30811/+5221
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294430 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Analog lines do not transfer CONNECTED LINE or execute the interception macros.rmudgett5-328/+342
Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294349 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Merged revisions 294312 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08 Nov 2010) | 1 line add missing unlock not present in 294277 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294313 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Merged revisions 294277 via svnmerge from jpeeler4-1/+27
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294278 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Merged revisions 294242 via svnmerge from mnicholson1-31/+40
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294243 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Set a default waittime, and make sure to convert it to millisecondstwilson2-4/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294207 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08valgrind reported references to freed memory during a mISDN hangup collision.rmudgett1-173/+188
Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294125 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Fixed deadlock avoidance issues while locking channel when adding thebbryant1-18/+10
Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294084 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Corret spelling and exampletwilson1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294049 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Tell people to use the correct common name for clients as welltwilson1-5/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294047 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Merged revisions 293969 via svnmerge from sruffell1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293970 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-04Fixes ringback tone on sip semi-attended transfer.dvossel1-0/+4
ABE-2168 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-04Do not output port in IPaddress for AMI sippeers.pabelanger1-1/+1
(closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293887 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Merged revisions 293806 via svnmerge from rmudgett2-17/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293807 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Avoid valgrind warnings for ast_rtp_instance_get_xxx_addresstwilson3-13/+66
The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293803 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293723 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293724 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293647 via svnmerge from rmudgett2-4/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02If manager and tls are disabled, do not display TCP/TLS Bindaddress.pabelanger1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293611 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Analog 3-way call would not connect all parties if one was using sig_pri.rmudgett3-30/+49
Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293530 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Use ast_sockaddr_from_sin function not memcpypabelanger1-3/+6
This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Only offer codecs both sides support for directmediatwilson1-7/+17
When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293493 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293417 via svnmerge from rmudgett2-24/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293418 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293340 via svnmerge from rmudgett2-23/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293341 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-29Modify sip_setoption to not complain about unknown options.jpeeler1-1/+0
This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293305 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28Merged revisions 293195-293196 via svnmerge from tilghman6-588/+1148
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ r293196 | tilghman | 2010-10-28 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293197 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28Merged revisions 293158 via svnmerge from jpeeler1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293159 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26Merged revisions 293118 via svnmerge from jpeeler1-21/+91
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293119 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26No need to define the struct if there are no users.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293081 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26Allow the DAHDI driver to compile, even with a sufficiently older version of ↵rmudgett4-11267/+11817
libpri. Fixes our Bamboo builds. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293046 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Several more defines that need to be altered for compiling against an older ↵tilghman1-4/+4
version of libpri git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Allow the DAHDI driver to compile, even with a sufficiently older version of ↵tilghman4-7719/+32795
libpri. Fixes our Bamboo builds. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292906 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Merged revisions 292867 via svnmerge from dvossel1-181/+225
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292868 f38db490-d61c-443f-a65b-d21fe96a405b
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