aboutsummaryrefslogtreecommitdiffstats
AgeCommit message (Collapse)AuthorFilesLines
2010-03-18Update new Local channel documentation.lmadsen1-34/+55
The original reporter, Kobaz, of an issue with a Local channel that inspired the Local channel documentation provided some tweaks to the documentation after testing what I had written. Hopefully anything that was vague or unclear has been cleaned up by these changes. (closes issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) Tested by: kobaz, lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253252 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-17Revert API change in release branchestwilson6-10/+10
This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-17Add french snipset to say.conf.lmadsen1-0/+31
Add the french snipset to say.conf. (Closes issue #15799) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253018 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16Backport chan_sip build fix for Mac OSX 10.6 from trunk.russell1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252928 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16Use uname -s, as done in trunk.russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252927 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16Apply codec_gsm Mac OS X 10.6 build fix that is in trunk and 1.6.X.russell1-5/+12
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252851 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16Don't treat warnings as errors for muted.russell1-0/+1
muted supports OS X, but uses functions marked as deprecated in 10.6. However, the functions are still supported, so just ignore the warnings for now and allow the build to proceed. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252766 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16Additional extensions.ael global variable fixes.lmadsen1-7/+7
Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252761 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Uh, yeah. Umask. I'm stupid.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252617 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Update extensions.ael file to not overlap extensions.conf.lmadsen1-10/+18
Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252533 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Revert last commit that had bad changed to configure.lmadsen2-20/+12
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252532 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Update extensions.ael file to not overlap extensions.conf.lmadsen2-12/+20
Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252531 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Typotilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252366 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Launch Asterisk on Mac OS X with launchd.tilghman3-1/+50
Reviewboard: https://reviewboard.asterisk.org/r/551/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252361 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-13Merged revisions 252089 via svnmerge from twilson9-51/+76
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252175 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Forward declaring dahdi_pri was already done.rmudgett1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251997 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Make chan_dahdi wakeup_sub() prototype not conditional.rmudgett1-5/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251986 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-09Use ast_strlen_zero to avoid a crash when a Dial() string isn't passed to ↵seanbright1-1/+1
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/branches/1.4@251410 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-08Fix Debian init script to not use -c.lmadsen1-1/+1
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/branches/1.4@251309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05Fix not being able to specify a URL in MOH class directory.jpeeler1-1/+1
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/branches/1.4@250786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Update existing Local channel documentation.lmadsen1-32/+405
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/branches/1.4@250613 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Make sure to clear red alarm after polarity reversal.jpeeler1-0/+12
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/branches/1.4@250480 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03fixes problem with duplicate TXREQ packetsdvossel1-3/+8
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/branches/1.4@250394 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. (closes issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update documentation to clarify purpose of unanswered option.lmadsen1-0/+6
(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/branches/1.4@250043 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update documentation to not imply we support overriding options.lmadsen1-13/+21
(issue #16855) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250041 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02revert ability to exit echo appalecdavis1-17/+10
caused a regression, as only supported VOICE, not VIDEO etc. Left in small formatting change. (issue #16880) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249946 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes ability to exit echo appalecdavis1-10/+18
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249845 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Fix crash in app_voicemail related to message counting.seanbright1-1/+1
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/branches/1.4@249671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Modify queued frames from local channels to not set the other side to upjpeeler1-30/+1
In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249536 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27overlap receiving: automatically send CALL PROCEEDING when dialplan startsalecdavis1-1/+14
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis (closes issue #16789) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249365 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27add a reference to the now-published IAX2 RFCkpfleming1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249234 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26For T.38 reINVITEs treat a 606 the same as a 488.mmichelson1-0/+2
(closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249100 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Ensure that monitor recordings are written to the correct location (again)jpeeler1-3/+3
This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248860 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Some platforms clear /var/run at boot, which makes connecting a remote ↵tilghman1-0/+12
console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248859 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Ensure that monitor recordings are written to the correct location.jpeeler1-8/+8
Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-24Make deletion of temporary greetings work properly with IMAP_STORAGEjpeeler1-4/+6
This same patch was merged in 220833, but was skipped in this branch erroneously. (closes issue #16170) Reported by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248668 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-24Remove color code sequences from verbose messages that go to logfiles.tilghman1-1/+1
(closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248582 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-23fixes invite with replaces deadlockdvossel1-14/+46
(closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248396 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-22Don't log to debug unless debug is turned onoej1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248268 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-20Make sure we support RTCP compound messages with zero reportsoej1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248106 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Backport crash fix from trunk to 1.4, whereby 'core show gracefully' could ↵tilghman1-17/+30
crash Asterisk. (closes issue #16470) Reported by: kjotte git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248012 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revision 247904 fromrmudgett1-27/+5
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247910 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Copy the calling party's account code to the called party if they don't ↵mnicholson1-0/+3
already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247651 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Add additional link to best practices document per jsmith.lmadsen1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247508 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Add best practices documentation.lmadsen1-0/+292
(issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247502 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Tweak argument handling for wget in the sounds Makefile.russell2-2/+3
1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247422 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Make sure that when autofill is disabled that callers not in the front of ↵mmichelson1-3/+5
the queue cannot place calls. (closes issue #16834) Reported by: kebl0155 Patches: app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247168 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-15Make the menuselect instructions correct by allowing 'make menuselect' to ↵tilghman1-0/+2
actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246709 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-12lock channel during datastore removaldvossel1-0/+2
On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246545 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-29 05:06:16 +0000