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2007-11-15Undoing previous commit since I realize it was wrongmmichelson1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89288 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Adding a missing mutex unlock.mmichelson1-0/+1
(closes issue 11256, reported and patched by ys) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89286 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Don't send re-invites during pending INVITE transactions.oej1-6/+19
Patch by one47 - thanks! Closes issue #9305 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89281 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Improve support for multipart messages. Code by gasparz, changesoej1-18/+25
by me (mostly formatting). Thanks, gasparz! Closes issue #10947 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89280 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14When a recording ends with '#', we are improperly trimming an extra 200ms ↵tilghman1-3/+10
from the recording. Reported by: sim Patch by: tilghman Closes issue #11247 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89275 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14Return the proper value when the srv_callback function executes properly.file1-1/+1
(closes issue #11240) Reported by: jtodd git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89260 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Fix building on newer systems which require a third arg to open() when using ↵qwell2-2/+2
O_CREAT. Issue 11238, reported by puzzled. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89254 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Revert change from revision 67064.qwell1-1/+1
It is documented behavior that if a parking extension already exists while using PARKINGEXTEN, dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you must keep this in mind, and handle the failure yourself. Issue 11237, reported by jon. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89248 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13If we set a value for qualify, we should actually pay attention to it, ↵tilghman1-2/+2
instead of overriding the value git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89246 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Reverting commit made in revision 89205 since it is unnecessary.mmichelson1-2/+2
Thanks to Kevin for pointing this out git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89241 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Debugging is running into the 16-lock limit. Increase to avoid.tilghman1-1/+1
(This define is only effective when debugging is turned on, so there's no effect for most installations.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89239 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Some sanity checking for MixMonitor. If only 1 argument is given, then the ↵mmichelson1-2/+2
args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89205 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Fix a typo pointed out by De_Mon on #asterisk-devqwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89194 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12If two config writes collide, file corruption could result. Use a mkstemp() ↵tilghman1-9/+22
file, instead. Reported by: paravoid Patch by: tilghman Closes issue #10781 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89191 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Fix two cases of memory corruption caused by background threads.tilghman2-0/+10
Reported by: atis Patch by: tilghman Fixes issue #10923 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12if we're NT and no number was dialed and overlapdial is set, we wait for the ↵crichter2-0/+18
ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12added restart all interfaces Restart_Indicator, to automatically send a ↵crichter4-29/+41
RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12fixed a state/event issue with overlapdial=yes when no extension matched. ↵crichter2-32/+44
removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12fixed the support for CW and therefore for the reject_cause option.crichter1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89170 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on ↵crichter6-2/+34
a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89169 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Properly say the seconds here..qwell1-1/+1
Issue 11203, fix described by vma. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89125 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Rework of the commit I made yesterday to use the already built-inmmichelson1-27/+14
ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89119 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Avoid warnings on load when using sample configuration files.qwell1-1/+1
Issue 11195, patch by eliel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89115 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08I made this same adjustment in trunk to fix a bug, and it makes sense to do ↵mmichelson1-1/+1
it in 1.4 as well. If an imapfolder is specified in voicemail.conf, don't ever explicitly connect to INBOX since it may not exist. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89111 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08fix a glaring bug in the new SRV record handling that would cause incorrect ↵kpfleming1-0/+1
weight sorting git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89105 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Typotilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89103 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Do not add a sip: to the beginning of the To URI unless needed.file1-1/+1
(closes issue #10756) Reported by: goestelecom git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89101 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Improve the devicestate logic for multiple devices. If any are available ↵file1-6/+15
then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89099 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Add support for allowing one outgoing transaction. This means if a response ↵file1-2/+5
comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89097 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07If callerid is configured in sip.conf use that for checking the presence of ↵file1-2/+2
an extension in the dialplan. (closes issue #11185) Reported by: spditner git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89095 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07The member refcount must be incremented, to avoid using it after deallocation.tilghman1-0/+6
A huge thanks go to lvl- for patiently providing the necessary valgrind output that was necessary to finding this problem of memory corruption. Reported by: lvl- Patch by: tilghman Closes issue #11174 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89093 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07This patch makes it possible for SIP phones to dial extensions defined with ↵mmichelson1-2/+24
'#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89090 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07In response to 10578, I just ran 1.4 thru valgrind; some of the config ↵murf3-4/+10
leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89088 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Fixing a segfault in the manager "core show channels concise" command.mmichelson1-1/+1
(closes issue #11183, reported by arnd and patched by ys) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89085 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Suppress AEL warnings on load.tilghman1-1/+2
Reported by: eliel Patch by: eliel Closes issue #11178 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89079 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Fix init_classes() so that classes that actually do have files loaded aren'trussell1-1/+1
treated as empty, and immediately destroyed ... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89053 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Correctly set the total number of channels from a zaptel transcoder board.qwell1-2/+1
SPD-49, patch by Matthew Nicholson. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89046 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06We went to the trouble of creating a method of tracking failed trylocks, ↵tilghman1-1/+1
then never turned it on (oops). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89045 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Bug fixes to tdd support in zaptel. oej1-16/+24
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89042 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06If someone were to delete the files used by an existing MOH class, and thenrussell1-3/+15
issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue #10948) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89037 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06closes issue #8786 - where the [catname](!) and ↵murf1-7/+66
[catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89036 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Make it so that if a peer is determined to be unreachable using qualify ↵file1-1/+1
their devicestate will report back unavailable. (closes issue #11006) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89032 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Fix improbable but possible memory leaks in chan_zap.file1-2/+10
(closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88994 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Remove some checks to see if locks are initialized from the non-DEBUG_THREADSrussell1-67/+0
versions of the lock routines. These are incorrect for a number of reasons: - It breaks the build on mac. - If there is a problem with locks not getting initialized, then the proper fix is to find that place and fix the code so that it does get initialized. - If additional debug code is needed to help find the problem areas, then this type of things should _only_ be put in the DEBUG_THREADS wrappers. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88931 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06update comment to match the state of the codekpfleming1-4/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88862 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Reworked deadlock avoidance in __ast_read. Restored audio to mmichelson1-20/+9
callback agents. (closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88826 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05After seeing crashes related to channel variables, I went looking around at therussell2-11/+61
ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88805 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05When traversing the list of channel variables here in transmit_invite(), the russell1-2/+8
asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Fix up some indentation.russell1-26/+26
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88765 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Merge changes from asterisk/team/kpfleming/SRV-priority-handlingrussell2-44/+150
Previously, the SRV record support in Asterisk was broken. There was no guarantee on what record Asterisk would choose to actually use. This set of changes improves the situation by ensuring that Asterisk will choose the highest priority record. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88719 f38db490-d61c-443f-a65b-d21fe96a405b
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