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certified/1.8.11-cert1
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certified/11.2-cert1
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certified/11.6-cert6
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certified/13.1-cert1
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certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
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certified/13.1-cert3
${ noResults }
13400 Commits (9eaaa987fc286297fcf37caf8f16ed63ff189b54)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
9eaaa987fc |
If fileconfig limit exceeds our maximum, then set the limit to the maximum.
(Closes issue #14888) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
3c027b6378 |
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
9b0c55768a |
Support "signaling" in addition to "signalling".
The sample configuration file has references to both spellings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
de69468fc2 |
Add lastms column to the contributed table designs
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
a4e46eb871 |
Handle a SIP race condition (reinvite before an ACK) properly.
RFC 5047 explains the proper course of action to take if a reINVITE is received before the ACK from a previous invite transaction. What we are to do is to treat the reINVITE as if it were both an ACK and a reINVITE and process it normally. Later, when we receive the ACK we had been expecting, we will ignore it since its CSeq is less than the current iseqno of the sip_pvt representing this dialog. (closes issue #13849) Reported by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson (license 60) Tested by: mmichelson, klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
200db93157 |
Oops, typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
34672a3919 |
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
1cb43cfa75 |
Permit zero-length text messages in SIP.
(Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
6f919c9851 |
Oops, missed this file in the last commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
a8dc553099 |
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
19918c7274 |
Backport resolution for file descriptor leak in 1.6.0 to 1.4.
This fixes short reads in http manager sessions, such as those done by the ast-gui branch. (Fixes AST-198) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187209 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
8e31a331b4 |
Fix a crash due to too few arguments to RetryDial.
(closes issue #14852) Reported by: junky Patches: retry_fix.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187135 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
89d04eef98 |
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
6562e1025c |
Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
3c97fd86c5 |
Fix a few typos of the word "frequency."
(closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
c8cdf25ccb |
Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.
Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
56e7ca00c1 |
Fix Macro documentation to match current (and intended) behavior.
(See -dev mailing list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
8c021d9334 |
Ensure that \r\n is printed after the ActionID in an OriginateResponse.
(closes issue #14847) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
e2c30564bf |
Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
|
89e29bfdc9 |
Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested
Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
3d9585f4ba |
Found a conflict in the last commit, due to multiple targets
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186445 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
6d08bad538 |
Distinguish in a sent email between simple sends and forwards.
(closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
ddb260532b |
Fix a problem with the crypto variable definitions not actually being defined properly.
(closes issue #14804) Reported by: jvandal git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
f3dd49684f |
Fix a memory leak in cdr_radius.
I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
09490bb688 |
Fix instructions in one-step parking comment to make more sense.
Changed a capital K to a lowercase k. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
|
6dbc379d01 |
ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
24fa699663 |
Merged revisions 186056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
0487d30a98 |
Avoid multiple warning messages in SIP, due to this column not existing
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
|
090f081e41 |
the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.
this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
36c92eec0e |
Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491
Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
b66bb8945d |
Fix a case where DTMF could bypass audiohooks.
This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
7cc775e1ed |
Fix crash that would occur if an empty member was specified in queues.conf.
(closes issue #14796) Reported by: pida git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
36a68f792e |
Use AST_SCHED_DEL_SPINLOCK instead of manually using the logic.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
e2eaf7f862 |
Fix Russian voicemail intro to say the word "messages" properly.
(closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Brooks
|
a2933fefef |
Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
c48480f5e4 |
Fix some state_interface stuff that was in trunk but not in the backport to 1.4.
Issue #14359 was fixed between the time that I posted the review of the backport of the state interface change for 1.4. This merges the changes from that issue back into 1.4. (closes issue #14359) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
3b1fcc38f1 |
Fix crash when moving audiohooks between channels.
Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185196 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
|
63ca43071e |
Update the channel allocation method documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Richard Mudgett
|
76bfa1d8ef |
Make chan_misdn BRI TE side normally defer channel selection to the NT side.
Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
fd7d3271c6 |
Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked.
(This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
cf7131dd6a |
Backport state interface changes to app_queue from trunk.
After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
df192b77df |
Improve our handling of T38 in the initial INVITE from a device.
We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
23de13f3da |
Ensure targs variable is fully initialized.
(closes issue #14758) Reported by: tim_ringenbach git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
0f37862a87 |
Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
|
3433f4ecef |
use new, improved 8kHz prompts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
d64575ac5c |
pri loop TestClient/TestServer fails: server SEND DTMF 8
app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Eliel C. Sardanons
|
a81abfbd02 |
Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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4a209dbe40 |
Change NULL pointer check to be ast_strlen_zero.
The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184078 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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38934ec0d0 |
Additionally note that the operator option needs an 'o' extension.
(Related to issue #14731) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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3e474b3b75 |
Fix a memory leak in res_monitor.c
The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |