Commit graph

3601 commits

Author SHA1 Message Date
Tilghman Lesher
423d4b8278 When GOSUB is invoked within an AGI, it may not exit correctly.
(closes issue #16216)
 Reported by: atis
 Patches: 
       20091110__atis_work.diff.txt uploaded by tilghman (license 14)
 Tested by: atis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-10 21:22:50 +00:00
Matthew Nicholson
aabff54c4b Add the 'relative-periodic-announce' option to app_queue to allow for calculating the time of announcments from the end of the previous announcment rather than from the beginning.
(closes issue #15260)
Reported by: tonils


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-09 16:28:31 +00:00
Tilghman Lesher
bcb09043b8 Yet another error message in the dialplan (thanks, rmudgett/russellb)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05 22:12:45 +00:00
Tilghman Lesher
8b447d9063 MEETME_INFO should not return a literal error message to the dialplan.
(closes issue #15450)
 Reported by: JimVanM
 Patches: 
       meetmeinfopatch.diff.txt uploaded by dbrooks (license 790)
 Tested by: JimVanM


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05 21:24:21 +00:00
Jeff Peeler
6aaa119e84 Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.

(closes issue #16167)
Reported by: marhbere



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05 21:23:06 +00:00
Tilghman Lesher
8ee06c6c3b Don't crash if no arguments are passed.
(closes issue #16119)
 Reported by: thedavidfactor


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-05 17:08:02 +00:00
Matthew Nicholson
317435a932 Added the 'a' option to app dial and modified app_dial to set the answertime when the called channel answers.
This change causes answertime to be correct even if the called channel hangs up during an announcement triggered by the A() option.

(closes issue #15936)
Reported by: falves11
Patches:
      dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
      dial-caller-answer1.diff uploaded by mnicholson (license 96)
Tested by: falves11, mnicholson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 21:39:33 +00:00
Matthew Nicholson
ed2ed2717a Merged revisions 227827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines
  
  This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements.
  
  (closes issue #16005)
  Reported by: falves11
  Patches:
        dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson, falves11
  
  Review: https://reviewboard.asterisk.org/r/407/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 21:03:33 +00:00
Tilghman Lesher
d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Tilghman Lesher
206d2cbc16 Don't crash when state_interface is NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 03:15:10 +00:00
Matthew Nicholson
7ed425ec80 This patch adds a sequence field to CDRs that can be combined with the linkedid or uniqueid field to uniquely identify a CDR.
(closes issue #15180)
Reported by: Nick_Lewis
Patches:
      cdr-sequence10.diff uploaded by mnicholson (license 96)
Tested by: mnicholson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03 21:21:09 +00:00
Joshua Colp
2263ced9dd Add support for using a hint when configuring a state interface using the format hint:<extension>@<context>.
(closes issue #15168)
Reported by: p_lindheimer
Patches:
      queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03 21:16:14 +00:00
Leif Madsen
f457dfecd1 Change warning message to debug message.
app_controlplayback outputs a warning, when in fact it is normal.

(closes issue #16071)
Reported by: atis
Patches:
      controlplayback_warning.patch uploaded by atis (license 242)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03 19:48:53 +00:00
Joshua Colp
7a17d87740 Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
  
  Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
  while the called party had not yet answered.
  
  This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
  file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
  
  (closes issue #14674)
  Reported by: ulogic
  Patches:
        bug14674.patch uploaded by jpeeler (license 325)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02 18:08:54 +00:00
David Vossel
776a14386a SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
  Connection setup takes awhile and before this it was being
  done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread:  Through the
  use of a packet queue and an alert pipe, the TCP helper thread
  can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
  up the tcptls_session lock.  This lock has been completely removed
  from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak:  When creating a tcptls_client the tls_cfg was alloced
  but never freed unless the tcptls_session failed to start.  Now the
  session_args for a sip client are an ao2 object which frees the
  tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
  of a client's ast_tcptls_session_args was done on the stack and
  stored as a pointer in the newly created tcptls_session.  Depending
  on the events that followed, there was a slight possibility that
  pointer could have been accessed after the stack returned.  Given
  the new changes, it is always accessed after the stack returns
  which is why I found it.

Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
  functions.  One for creating and allocating the new tcptls_session,
  and a separate one for starting and handling the new connection.
  This allowed me to create the tcptls_session, launch the helper
  thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
  This is done by using an alert pipe to wake up the thread if new
  data needs to be sent.  The thread's sip_threadinfo object contains
  the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
  accessed outside of the helper thread for every write (queuing of a
  packet).  For easy lookup, I moved the threadinfo objects from a
  linked list to an ao2_container.

(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys

(closes issue #15894)
Reported by: dvossel
Tested by: dvossel

Review: https://reviewboard.asterisk.org/r/380/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22 19:55:51 +00:00
Tilghman Lesher
d9f72c1893 Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276)
 Reported by: klaus3000
 Patches: 
       app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65)
 Tested by: jamesgolovich


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22 19:10:04 +00:00
Tilghman Lesher
496282194c Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
  
  Fix documentation for ast_softhangup() and correct the misuse thereof.
  (closes issue #16103)
   Reported by: majorbloodnok
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22 17:11:23 +00:00
Kevin P. Fleming
cdd1f9e296 Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.

During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.

Review: https://reviewboard.asterisk.org/r/379/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 21:08:47 +00:00
Tilghman Lesher
0776bcff64 Apparently, I don't need to specify the ".so" suffix to get a match
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 15:42:47 +00:00
Tilghman Lesher
a2f809c127 Turn on DENOISE filter for all conference participants.
(Fixes SWP-238)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 15:21:30 +00:00
Joshua Colp
b7a50aeddc Merged revisions 224565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines
  
  Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
  
  (closes issue #14763)
  Reported by: cupotka
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19 19:49:09 +00:00
Tilghman Lesher
8f9edddc27 Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's.
This corrects an issue reported on the -users list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19 00:05:56 +00:00
Jeff Peeler
a39f3a7521 Readd removed ability to allow listening to one side of the call in app_chanspy
(Option o)

(closes issue #15675)
Reported by: john8675309
Patches:
      issue15675patchtrunk.txt uploaded by dbrooks (license 790)
Tested by: jgutierrez on users list:
 http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15 15:57:14 +00:00
Terry Wilson
1f9f1562ba Revert inadvertant code commit to app_originate
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13 01:58:09 +00:00
Terry Wilson
a8034cd770 Fix handling of notification calls w/ the dialing api
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13 01:51:46 +00:00
Jeff Peeler
832be82dfb Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines
  
  Ensure ringing continues for branched calls after progress is received
  
  While waiting for an answer, don't send progress for branched calls
  for which ringing was sent.
  
  (closes issue #15028)
  Reported by: fnordian
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12 23:48:09 +00:00
Kevin P. Fleming
e197f85b8c Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.

(closes issue #16025)
Reported by: jamicque


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12 14:25:29 +00:00
Kevin P. Fleming
7c71e98879 Initiate T.38 switchover when acting as called party, regardless of FAX direction.
SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
to initiate a switchover to T.38, not the direction that the FAX transfer will
take place.

(closes issue #16039)
Reported by: jamicque


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09 20:58:44 +00:00
Mark Michelson
66e993de95 Fix potential memory leaks.
ABE-1998



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09 18:13:57 +00:00
Kevin P. Fleming
1c9fe00920 Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
  
  Fix ao2_iterator API to hold references to containers being iterated.
  
  See Mantis issue for details of what prompted this change.
  
  Additional notes:
  
  This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
  has become an enum instead of a macro, with a name that fits our
  naming policy; also, it is now necessary to call
  ao2_iterator_destroy() on any iterator that has been
  created. Currently this only releases the reference to the container
  being iterated, but in the future this could also release other
  resources used by the iterator, if the iterator implementation changes
  to use additional resources.
  
  (closes issue #15987)
  Reported by: kpfleming
  
  Review: https://reviewboard.asterisk.org/r/383/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-06 01:24:24 +00:00
Matthias Nick
00bb578898 Prevents from division by zero
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30 21:15:01 +00:00
Sean Bright
0e805a51ec Modify VoiceMailMain()'s a() argument to allow mailboxes to be specified by name.
(closes issue #14740)
Reported by: pj
Patches:
      issue14740_09022009.diff uploaded by seanbright (license 71)
Tested by: seanbright, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30 15:11:21 +00:00
Sean Bright
3ac28f1e0f Clarify documentation for VoiceMailMain()'s a() option.
We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
      __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30 14:47:58 +00:00
Matthew Nicholson
4b4432b67e Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel.
(closes issue #15964)
Reported by: shrift


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-29 19:49:02 +00:00
Jeff Peeler
a9154a905a Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!

(closes issue #14949)
Reported by: noahisaac
Patches:
      vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), 
      modified by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-29 16:58:29 +00:00
Jeff Peeler
f150b48bc0 Add bridge related dial flags to the bridge app
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.

(closes issue #13165)
Reported by: tim_ringenbach
Patches:
      app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
      modified by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24 20:29:51 +00:00
Tilghman Lesher
1cf5422dc8 Merged revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
  
  Implicitly sending a progress signal breaks some applications.
  Call Progress() in your dialplan if you explicitly want progress to be sent.
  (Reverts change 216430, closes issue #15957)
  Reported by: Pavel Troller on the Asterisk-Dev mailing list
  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24 19:41:02 +00:00
Tilghman Lesher
9298b2602a Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward.
(closes issue #15739)
 Reported by: DLNoah, jeffg
 Patches: 
       20090914__issue15739.diff.txt uploaded by tilghman (license 14)
       20090922__issue15739.diff.txt uploaded by tilghman (license 14)
 Tested by: DLNoah, jeffg


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24 07:39:44 +00:00
Tilghman Lesher
6c3a3dabe0 Merged revisions 219816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
  
  When IMAP variables were changed during a reload, Voicemail did not use the new values.
  This change introduces a configuration version variable, which ensures that
  connections with the old values are not reused but are allowed to expire
  normally.
  (closes issue #15934)
   Reported by: viniciusfontes
   Patches: 
         20090922__issue15934.diff.txt uploaded by tilghman (license 14)
   Tested by: viniciusfontes
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-22 21:43:22 +00:00
Tilghman Lesher
340d4b1a93 Missing value setting line for maxsecs/maxmessage
(closes issue #15696)
 Reported by: fhackenberger
 Patches: 
       maxsecs.patch uploaded by fhackenberger (license 592)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-18 13:54:51 +00:00
Sean Bright
a48d489568 Get this compiling under dev-mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-17 16:25:38 +00:00
Tilghman Lesher
cd88adfc6a Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
 Reported by: junky
 Patches: 
       __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
 Tested by: amilcar, junky, flujan, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-17 00:58:10 +00:00
Tilghman Lesher
e876206693 Merged revisions 218730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines
  
  If the user enters the same password as before, don't signal an error when the change does nothing.
  (closes issue #15492)
   Reported by: cbbs70a
   Patches: 
         20090713__issue15492.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-15 22:33:10 +00:00
Tilghman Lesher
95da50292e Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines
  
  Ensure FollowMe sets language in channels it creates.
  Also, not in the original bug report, but related fields are accountcode and
  musicclass, and the inheritance of datastores.
  (closes issue #15372)
   Reported by: Romik
   Patches: 
         20090828__issue15372.diff.txt uploaded by tilghman (license 14)
   Tested by: cervajs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-15 16:04:41 +00:00
Tilghman Lesher
a873ad7a9b Recorded merge of revisions 218331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines
  
  Don't say "Please try again" if we don't give the user another chance to try again.
  (issue #15055, SWP-129)
   Reported by: jthurman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-14 19:29:48 +00:00
Matthew Nicholson
3b2023290f Merged revisions 218223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines
  
  Ensure we don't pickup ourselves when doing pickup by exten.
  
  (closes issue #15100)
  Reported by: lmsteffan
  Patches:
        (modified) pickup.patch uploaded by lmsteffan (license 779)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-14 14:57:23 +00:00
Tilghman Lesher
85f18fcb8f Merged revisions 217989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines
  
  Don't ring another channel, if there's not enough time for a queue member to answer.
  (Fixes AST-228)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10 23:54:51 +00:00
Sean Bright
245b163755 Fix compilation of app_meetme.
Reported by ebroad in #asterisk-bugs


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08 22:17:08 +00:00
Tilghman Lesher
555ed0464f Merged revisions 217156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines
  
  When MOH is playing on the channel, announcements sent through the conference are not heard.
  (closes issue #14588)
   Reported by: voipas
   Patches: 
         20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
   Tested by: lmadsen, twisted, tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08 20:28:41 +00:00
Sean Bright
40d83cf748 Use ast_free() instead of free().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 19:29:02 +00:00