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${ noResults }
13980 Commits (fa3c085c7b4e4899a0ac93ad98f8879c048fbba4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Leif Madsen
|
d03a21d5f8 |
Revert last commit that had bad changed to configure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
0434ec7ad6 |
Update extensions.ael file to not overlap extensions.conf.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@252531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Tilghman Lesher
|
aac3bf7298 |
Typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
abe8ae27a3 |
Launch Asterisk on Mac OS X with launchd.
Reviewboard: https://reviewboard.asterisk.org/r/551/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Terry Wilson
|
529e8af144 |
Merged revisions 252089 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Richard Mudgett
|
a247e69d65 |
Forward declaring dahdi_pri was already done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@251997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
0bc0edcda9 |
Make chan_dahdi wakeup_sub() prototype not conditional.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@251986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Sean Bright
|
85f79116ac |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@251410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
dfd1365321 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@251309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jeff Peeler
|
093e1d34f3 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
db883a27bd |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
3896fecd58 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
David Vossel
|
8dbeb5925d |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
9ff28fd6e8 |
Update IMAP documentation.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
19c43ed644 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@250043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
c91785154c |
Update documentation to not imply we support overriding options.
(issue #16855) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@250041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Alec L Davis
|
3d3116e656 |
revert ability to exit echo app
caused a regression, as only supported VOICE, not VIDEO etc. Left in small formatting change. (issue #16880) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@249946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Alec L Davis
|
63e614d896 |
fixes ability to exit echo app
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@249845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Sean Bright
|
daf062ece8 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@249671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
9a4e4eb749 |
Modify queued frames from local channels to not set the other side to up
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@249536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Alec L Davis
|
271908886f |
overlap receiving: automatically send CALL PROCEEDING when dialplan starts
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@249365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Kevin P. Fleming
|
62f453c9d7 |
add a reference to the now-published IAX2 RFC
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@249234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Mark Michelson
|
99e1a0f967 |
For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@249100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
dbdbc92a4a |
Ensure that monitor recordings are written to the correct location (again)
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@248860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
096a41e440 |
Some platforms clear /var/run at boot, which makes connecting a remote 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@248859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
3371a165f5 |
Ensure that monitor recordings are written to the correct location.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@248757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
e896cc88cf |
Make deletion of temporary greetings work properly with IMAP_STORAGE
This same patch was merged in 220833, but was skipped in this branch erroneously. (closes issue #16170) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
252704411c |
Remove color code sequences from verbose messages that go to logfiles.
(closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
8d9b578022 |
fixes invite with replaces deadlock
(closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Olle Johansson
|
ef93e3343e |
Don't log to debug unless debug is turned on
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Olle Johansson
|
b39082c470 |
Make sure we support RTCP compound messages with zero reports
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
afb69ed7c8 |
Backport crash fix from trunk to 1.4, whereby 'core show gracefully' could crash Asterisk.
(closes issue #16470) Reported by: kjotte git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
7b5a46027f |
Merged revision 247904 from
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@247910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Matthew Nicholson
|
815f726220 |
Copy the calling party's account code to the called party if they don't already have one.
(closes issue #16331) Reported by: bluefox Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@247651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
e34cdeceaf |
Add additional link to best practices document per jsmith.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@247508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
879ee0b9a6 |
Add best practices documentation.
(issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@247502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
|
e691e5c6a6 |
Tweak argument handling for wget in the sounds Makefile.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@247422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
a01809c2c2 |
Make sure that when autofill is disabled that callers not in the front of 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@247168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
ace50777f2 |
Make the menuselect instructions correct by allowing 'make menuselect' to 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@246709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
ff6dd92e3e |
lock channel during datastore removal
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@246545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jason Parker
|
403eb70599 |
Fix some silly formatting, and remove unnecessary option_debug checks
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@246460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
47de4d5211 |
fixes random deadlock in app_queue with use_weight during reload
(closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@246115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
793e58a924 |
Include examples of FILTER usage in extension patterns where a "." may be a risk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
b279512253 |
Make sure that res_smdi loads regardless of configuration, since chan_dahdi depends on res_smdi
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245909 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
5a823cad50 |
Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jason Parker
|
90f2840242 |
Remove reference of documentation in source directory.
People don't always build Asterisk from source (distro packages, anybody?). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
ed94dddd97 |
Res_features depends on res_adsi in 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
c540f2eff1 |
Remove contrib/firmware directory as it is empty
Remove explicit license for IAXy firmware as it is no longer included in the tree git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
|
90bf027119 |
Update main copyright date.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@244926 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
735485effc |
Change channel state on local channels for busy,answer,ring.
Previously local channels channel state never changed. This became problematic
when the state of the other side of the local channel was lost, for example
during a masquerade. Changing the state of the local channel allows for the
scenario to be detected when the channel state is set to ringing, but the peer
isn't ringing. The specific problem scenario is described in 164201. Although
this was noted on one of the issues, here is the tested dialplan verified to
work:
exten => 9700,1,Dial(Local/*9700@default&Local/#9700@default)
exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
exten => *9700,n,wait(3) ;3 works, 1 did not
exten => *9700,n,Dial(SIP/5001)
exten => #9700,1,Wait(1) ;1 works, 3 did not
exten => #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1)
(closes issue #14992)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@244785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
16 years ago |