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2010-03-25Don't remove local copies of utils in uninstall.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254800 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix DEBUG_THREADS issue with out-of-tree modules.qwell2-18/+57
Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254714 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Update Asterisk 1.4 to use menuselect trunk.russell1-6/+38
Review: https://reviewboard.asterisk.org/r/590/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Add doxygen for acl.hmmichelson1-0/+173
Review: https://reviewboard.asterisk.org/r/528 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254552 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Several fixes regarding RFC2833 DTMF detection.mmichelson1-13/+36
Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem Review: https://reviewboard.asterisk.org/r/558 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254452 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Handle new SRCCHANGE control message here tootwilson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254451 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Ensure that monitor recordings are written to the correct location (again)jpeeler1-22/+25
This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254235 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Revert revisions 254046 and 254098.qwell7-1006/+1103
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254161 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Add note about the out-of-tree module ABI changes.qwell1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254098 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Allow out-of-tree modules to load, regardless of ↵qwell6-1103/+999
DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This can be guaranteed by forcing the ABI to no longer change when these compiler flags are set. An unfortunate side-effect to this is that there is an ABI change here. However, there is some mitigation. Existing modules *will* fail to load since they would require functions that no longer exist. Review: https://reviewboard.asterisk.org/r/508/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254046 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-22Unconditionally copy the caller's account code to the called party.mnicholson1-3/+1
(related to issue #16331) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253799 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-21Fix final link on FreeBSD by adding the PTHREAD_CFLAGS.russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253670 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve a number of FreeBSD build issues.russell10-12/+15
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253631 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Typo found while fixing issue #16961.lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253349 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Synchronize text in localchannels.txt and localchannels.tex.lmadsen1-7/+7
(issue #16963) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253260 f38db490-d61c-443f-a65b-d21fe96a405b
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
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-01 04:15:07 +0000