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2008-07-10Removed the fn2 field from the vm_state structure.mmichelson1-13/+7
fn2 was used in three functions. In every case, it was initialized in the function it was used in. This meant there was no need to have it in a malloc'd structure just taking up space. Furthermore two of the functions it was used in were completely unnecessary since fn2 was set to exactly the same value as the vm_state's fn string. fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is 4096. This equates to a 4K memory savings per vm_state allocated. Since there is a vm_state malloc'd for every voicemail user on the system, this could potentially add up nicely if there are lots of users. In addition, a vm_state is allocated on the stack each time a caller calls the VoiceMailMain application, meaning that there is a significant stack savings with this patch too. Of course, a single vm_state struct still takes up approximately 20K on my system (when using IMAP storage. Without IMAP storage, there would be about another 300 bytes fewer usage), even with this removal. Further optimizations are probably possible, but most likely not as easy as this one. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129734 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Fixes a bug where the interface for a queue member gets reloaded as the ↵bbryant1-3/+3
state_interface, if a state_interface was set, on reload because the state_interface isn't stored in the ast_db. (closes issue #13043) Reported by: jvandal Patches: app_queue.patch uploaded by jvandal (license 413) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129684 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10A couple more minor text changesseanbright1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129642 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Remove extraneous \n. Pointed out by eliel on #asterisk-dev.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129638 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Remove deprecated 'show parkedcalls' CLI commandmvanbaak1-11/+1
(closes issue #13038) Reported by: eliel Patches: finish.deprecate.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129581 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Merged revisions 129567 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines Note that pbx_spool.so is the module used for call files (inspired by a question in #asterisk) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129569 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Blocked revisions 129505 via svnmergeseanbright0-0/+0
........ r129505 | seanbright | 2008-07-10 09:57:19 -0400 (Thu, 10 Jul 2008) | 2 lines Update svn:ignore ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129506 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-10Update svn:ignoreseanbright0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129503 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-09Merged revisions 129436 via svnmerge from mmichelson1-4/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129437 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-09Add Proceeding() application (#13025)mattf1-2/+18
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129399 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-09Merged revisions 129343 via svnmerge from seanbright4-55/+95
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines Look for the system installed awk instead of assuming it's at /usr/bin/awk. Pointed out by jmls via #asterisk-dev. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129344 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-09Code wasn't ready to be merged - see -dev list discussiontilghman7-58/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129307 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Fix compilation error when IMAP storage is enabledmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Blocked revisions 129208 via svnmergemmichelson0-0/+0
........ r129208 | mmichelson | 2008-07-08 16:31:51 -0500 (Tue, 08 Jul 2008) | 3 lines Update documentation to have the correct option name ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129210 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Blocked revisions 129158 via svnmergemmichelson0-0/+0
........ r129158 | mmichelson | 2008-07-08 16:09:10 -0500 (Tue, 08 Jul 2008) | 14 lines Backport TCP-related timeouts to IMAP voicemail in 1.4 since it should solve bugs people are experiencing. Specifically, there are times where communication with the IMAP server causes system calls to block forever. If this should happen when querying the mailbox so that chan_sip's do_monitor thread can send MWI to a phone, it means that SIP calls cannot be processed any more. The timeout options are outlined in doc/imapstorage.txt. Defaults for the timeouts are sixty seconds. (closes issue #12987) Reported by: mthomasslo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129159 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Fix a bug in SRV lookups where dnsmgr would discard everything but the bbryant3-6/+5
first SRV result from DNS before processing weights and priorities and dns_parse_answer wouldn't report that there were no records in DNS unless a failure occured. Also fixed a bug where dnsmgr_refresh would report that a entry was being changed when ast_gethostbyname had failed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129156 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 129149 via svnmerge from tilghman3-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129152 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Reduce length of time that 'asterisk -rx' waits.tilghman1-2/+10
(closes issue #13001) Reported by: eliel Patches: 20080708__bug13001.diff.txt uploaded by Corydon76 (license 14) 20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64) Tested by: Corydon76, eliel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129114 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 129047 via svnmerge from tilghman1-10/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines Timestamp decoding for video mini-frames is bogus, because the timestamp only includes 15 bits, unlike voice frames, which contain a 16-bit timestamp. (closes issue #13013) Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129048 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Janitor project to convert sizeof to ARRAY_LEN macro.bbryant19-70/+77
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129045 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Update app_fax for better compatibility with spandsp 0.0.5. Add a call torussell1-7/+7
t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129006 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 128950 via svnmerge from oej1-9/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines Don't hangup the call if we can't resolve the Contact if there's a proxy route set for the call. ---- This comment was added a while ago and today it hit me badly. /* OEJ: Possible issue that may need a check: If we have a proxy route between us and the device, should we care about resolving the contact or should we just send it? */ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 128912 via svnmerge from oej1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably. Reported by: johan Patches: 12746.txt uploaded by oej (license 306) Tested by: johan (issue #12746) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128927 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 128856 via svnmerge from tilghman1-3/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines Check for non-NULL before stripping characters. (closes issue #12954) Reported by: bfsworks Patches: 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) Tested by: deti ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128857 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Merged revisions 128812 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines Stop using deprecated method, as requested by Kevin. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128830 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Merged revisions 128795 via svnmerge from russell1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines Fix handling of when a pvt disappears. Properly return the pvt locked and don't hold the pvt lock while destroying the ast_channel. (closes issue #13014) Reported by: jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128796 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Merged revisions 128737 via svnmerge from seanbright1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines Remove spurious trailing whitespace from log messages and fix a spelling error in a log message. (closes issue #13017) Reported by: jpgrayson Patches: chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492) chan_iax2_spelling.patch uploaded by jpgrayson (license 492) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128738 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Crapmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128733 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain,mmichelson1-2/+6
app_voicemail would attempt to play a file called vm-foo instead of playing vm-INBOX to play the "new" sound file. This commit fixes that issue. This may fix one of the problems reported in issue #12987 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128731 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Get app_voicemail compiling when IMAP storage is used.mmichelson1-12/+12
Brought up by reporter on issue #12987 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128690 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Merged revisions 128639 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines By using the iaxdynamicthreadcount to identify a thread, it was possible for thread identifiers to be duplicated. By using a globally-unique monotonically- increasing integer, this is now avoided. (closes issue #13009) Reported by: jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128640 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Blocked revisions 128637 via svnmergekpfleming0-0/+0
........ r128637 | kpfleming | 2008-07-07 11:51:50 -0500 (Mon, 07 Jul 2008) | 9 lines use tzafrir's patch to fix this problem properly... i made the previous set of changes without thoroughly testing them, doh! (closes issue #12911) Reported by: tzafrir Patches: custum_dahdi_configure_2.diff uploaded by tzafrir (license 46) Tested by: tzafrir ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128638 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Update a few instances of "extensions reload" to "dialplan reload"mmichelson2-2/+2
in the documentation. Patch provided by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128599 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07As pointed out on the -dev list, actually use the result of find_peer() so thatrussell1-2/+3
a peer reference is not leaked. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵oej2-6/+3
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Fixing issues with "sip show settings"oej2-16/+32
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Remove unused variable "expiry"oej1-3/+2
- Set global_outboundproxy.force at reload. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128491 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06More doxygen comments.oej1-37/+52
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128485 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Formatting changesoej1-10/+14
- Doxygen changes - Replacing a doxygen description that was copied from another function git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128455 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Adding note about incorrect manager registration...oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128454 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Adding documentation on the T.140 support in Asterisk. This is a function ↵oej1-0/+84
that we're the reference implementation on now. :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128417 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Remove comments that doesn't make sense. The deprecation of type=user will ↵oej1-2/+0
come at a later stage, as indicated by previous commit message git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128381 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Fix severe problem with my previous commit of "kill-the-user". Russell saw a ↵oej1-2/+2
problem with this code, but not the correct problem. Thanks, anyway! ;-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128379 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Changing name of global api call to ast_* oej4-5/+5
My mistake, pointed out by Russell. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128378 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Disabling code used by dumpdb with #ifdef, since I believe we might use it ↵oej1-6/+12
sometime in the future, but also want to avoid compiler warnings now. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128344 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06Removing the CLI dumpdb command (see asterisk-dev discussion and decision)oej1-284/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128343 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Adding a few remindersoej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128294 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Adding doxygen comments to missing parts, moving some #defineoej1-10/+27
...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128290 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Adding TCP and TLS to "sip show settings". oej1-6/+19
TLS needs to have one configuration per configured domain at some point. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128287 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Add some comments...oej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128284 f38db490-d61c-443f-a65b-d21fe96a405b
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