mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
13415 Commits (ef9ef40c19c0683eba4cdba7d1877ba7a47ca90d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
ef9ef40c19 |
Don't treat a NOANSWER like a CHANUNAVAIL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
5de742a483 |
Properly handle @s within hints in AEL.
AEL was not handling the case of a device hint containing an @ symbol, which caused parking hints (e.g. hint(park:exten@context)) to error out the parser. This patch makes AEL treat the @ the same way it treats colon and ampersand now, meaning the characters are included in verbatim. (closes issue #14941) Reported by: bpgoldsb Patches: bug14941.patch uploaded by seanbright (license 71) Tested by: bpgoldsb git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
100aa13ae2 |
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results. Change test of mmap results to compare against MAP_FAILED git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
df7bee29a5 |
Move the check for chan->fdno == -1 to after the zombie/hangup check.
Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
5405c62098 |
Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
55e28f890d |
Modifed/added some debug messages.
JIRA ABE-1835 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189134 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
f870165dbe |
Make Busy() application set the CDR disposition to BUSY.
(closes issue #14306) Reported by: cristiandimache git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
df2bc7d715 |
Fix a bug where a value used to create the channel name was bogus.
This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
bf5b92f004 |
Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been.
(issue AST-210) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188937 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
611cf94f90 |
Only update realtime, if global option rtupdate != false
(closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
4b6846a9dd |
Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone.
JIRA ABE-1835 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
caef916825 |
Umask should not be exported into global namespace.
(closes issue #14912) Reported by: jcapp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
4c6e1bd0a5 |
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
730f2882d3 |
Update ast_readvideo_callback to match ast_readaudio_callback.
This fixes potential refcount errors that may occur on ast_filestreams. AST-208 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
f967bdc670 |
audio_audiohook_write_list() does not correctly update sample size after ast_translate.
audio_audiohook_write_list() does not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. While no 16kz codecs are supported in 1.4 at the moment, this will save headaches in the future if they ever are. the sample size is now updated after translating to reflect this possibility. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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200db93157 |
Oops, typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |