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2010-03-10Reduce the amount of database access for HAVE_PRI_SERVICE_MESSAGES.rmudgett3-125/+113
Rework HAVE_PRI_SERVICE_MESSAGES to not use the active values directly from the database. Database access is likely expensive. Database access now only happens on initialization, destruction, and when the B channel is taken in or out of service. This change is not related to call waiting but it would cause the search for a call waiting interface to be very expensive and slow down D channel message servicing. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-09Build system modifications to ensure that Asterisk properly builds on Mac OS ↵tilghman2-0/+11
X 10.6. (closes issue #16997) Reported by: jquinn Patches: 20100309__issue16997__2.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251475 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-09Blocked revisions 251410 via svnmergeseanbright0-0/+0
........ r251410 | seanbright | 2010-03-09 14:29:39 -0500 (Tue, 09 Mar 2010) | 8 lines Use ast_strlen_zero to avoid a crash when a Dial() string isn't passed to ParkAndAnnounce (closes issue #16731) Reported by: sebele67 Patches: issue16731_20100129.diff uploaded by seanbright (license 71) Tested by: sebele67 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251411 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-08Merged revisions 251309 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010) | 13 lines Fix Debian init script to not use -c. When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. (closes issue #16784) Reported by: pabelanger Tested by: pabelanger, mnick, davidw, mutineer612 (closes issue #16887) Reported by: jlpedrosa Tested by: jlpedrosa, mutineer612 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251310 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-08Remove portions that weren't meant to be committed for the OS X compat fixtilghman4-28003/+7007
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251263 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-08Change needed to make Mac OS X 10.6 happytilghman6-7007/+28008
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-07Clean transmit_* for start/stop media transmission mvanbaak1-42/+33
Small patch changing skinny_set_rtp_peer to use transmit_stopmediatransmission and to use new transmit_startmediatransmission. Basic testing on 30VIP's by wedhorn Basic testing on 7960 by me (closes issue #16956) Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by wedhorn (license 30) Tested by: wedhorn,mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251222 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-07Cleanup transmit_callstate handling mvanbaak1-117/+132
Broke the various functions included in transmit_callstate to their own functions. Transmit_callstate now just transmits callstate. Generally left the functionality as it was, which highlight some minor code issues (eg multiple transmit_callstate's). I did however revise the hint code usage of the old transmit_callstate as it it not appropriate to put a device on hook based on the change of a hinted device. (closes issue #16939) Reported by: wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license 30) Tested by: mvanbaak,wedhorn git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251221 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-07small log issue from bug 0016664may1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251181 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-06Fix a crash in SIP blind transfer handling found by an automated external test.russell1-0/+9
The first real test added to the external test suite found a pretty nasty crash that occurred in Asterisk trunk. The crash was due to a race condition between the REFER handling and channel destruction in the channel thread. After the transfer has been completed, we go back to the transferrer channel and try to lock it so we can fire off a CEL event. However, there was no guarantee that the channel was still around at that point since it's racing against the channel thread. Since ast_channel is a reference counted object, the fix is simple. The code unlocks the transferrer channel before finally completing the transfer with an async goto. At this point the channel thread is going to start call tear down and the channel will eventually be destroyed. To ensure that the channel is valid when we want to fire off the CEL event, increase the channel's reference count. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251137 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05fixes xml error in func_pitchshiftdvossel1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251087 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05PITCH_SHIFT dialplan functiondvossel2-0/+505
The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251038 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Remove pbx_gtkconsole and related gtk1 checks.russell6-28341/+7028
Review: https://reviewboard.asterisk.org/r/541/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251022 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Fix app_followme playing wrong sound files.jpeeler1-7/+7
Fixes regression introduced in 140167 that uses the wrong variable names. (closes issue #16930) Reported by: ianc Patches: fix_reload_followme.diff uploaded by ianc (license 998) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250979 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Fix up some of chan_sip's usage of the RTP engine API.russell1-3/+43
The get_local_address() function for an RTP instance was used when building an SDP, but the results were not honored. The RTP engine activate() function was not being used once we have determined that media will now flow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Missing quote in ODBC query.tilghman1-1/+1
(closes issue #16953) Reported by: elguero Patches: app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250913 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Fix up the ast_rtp_property enum.russell1-2/+7
The mis-placement of the latest entry meant that when it was set, it was writing one index past the end of the properties array in the ast_rtp_instance (which happened to be the local_address field). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250871 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Merged revisions 250786 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04 Mar 2010) | 9 lines Fix not being able to specify a URL in MOH class directory. Don't attempt to chdir on a URL! (closes issue #16875) Reported by: raarts Patches: moh-http.patch uploaded by raarts (license 937) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250787 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-04Adjust XML for func_channel to indicate that rtpdest can take a "text" argument.mmichelson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250730 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Recorded merge of revisions 250613 via svnmerge from lmadsen0-0/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010) | 11 lines Update existing Local channel documentation. A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250614 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Update existing Local channel documentation.lmadsen1-57/+461
A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250609 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Removed cdrflags from ast_channel structure.rmudgett6-17/+0
Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250480 via svnmerge from jpeeler2-1/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250481 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Changes 0ms to <1ms in cli END results during 'test execute'dvossel1-2/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250478 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250394 via svnmerge from dvossel1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250395 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Add new config option to control AMI alarm event reporting in chan_dahdi.jpeeler3-17/+90
New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03One more fix to editlinetilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250346 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Eliminate remaining libedit warnings (shown in bamboo)tilghman4-0/+134
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250303 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Updated CHANGES file to mention res_fax and res_fax_spandsp.mnicholson4-2/+16
Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250302 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03fixes signed to unsigned int comparision issue for FaxMaxDatagram value.dvossel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250246 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03fixes assumption that test failed if it did not pass when generating resultsdvossel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250237 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02base64 unit testdvossel1-0/+54
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250235 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merge missed files from res_fax/res_fax_spandsp merge.mnicholson2-0/+273
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250213 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merge res_fax and res_fax_spandsp.mnicholson4-4/+2670
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02adds 'p' option to PickupChandvossel2-6/+62
The 'p' option allows the PickupChan app to pickup a ringing phone by looking for the first match to a partial channel name rather than requiring a full match. (closes issue #16613) Reported by: syspert Patches: pickipbycallid.patch uploaded by syspert (license 938) pickupbycallerid_v2.patch uploaded by dvossel (license 671) Tested by: dvossel, syspert git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250141 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update IMAP documentation.lmadsen1-0/+6
Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 250043 via svnmerge from lmadsen1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250045 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Recorded merge of revisions 250041 via svnmerge from lmadsen0-0/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (issue #16855) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250042 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update documentation to not imply we support overriding options.lmadsen1-12/+20
(closes issue #16855) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250037 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Fix literal values wrapped in documentation.lmadsen1-2/+3
(closes issue #16145) Reported by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02revert ability to exit echo appalecdavis1-20/+11
caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249947 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Add missing description of the PARKINGLOT variable in XML documentation.lmadsen1-0/+3
(closes issue #16743) Reported by: snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249925 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Convert some DUNDI functions to XML documentation.lmadsen1-23/+76
(closes issue #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by snuffy (license 35) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249912 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes adaptive jitterbuffer configurationdvossel24-13/+91
When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Fix several XML documentation validate errors.lmadsen3-51/+40
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249892 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fix build by checking result of symlink in test_voicemail_vmsaynamejpeeler1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249891 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Add new application VMSayName for use with voicemail.jpeeler2-3/+150
VMSayName that will play the recorded name of the voicemail user if it exists, otherwise will play the mailbox number. A unit test has been written to verify correct functionality called test_voicemail_vmsayname. (closes issue #14973) Reported by: ghjm Review: https://reviewboard.asterisk.org/r/530/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249889 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes ability to exit echo appalecdavis1-11/+20
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249801 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fix asterisk setting of pritimers from chan_dahdi.confalecdavis1-4/+5
regression since sig_pri split. (issue #16909) Reported by: alecdavis Patches: pritimer.asterisk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249759 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249671 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249672 f38db490-d61c-443f-a65b-d21fe96a405b
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