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r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines
- remove whitespaces between tags in received XML packets before giving
them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
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r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines
Merged revisions 111341 via svnmerge from
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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines
(closes issue #12302)
Reported by: pj
Tested by: murf
These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.
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r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.
(issue #12164, reported by milazzo and jsmith, patches by me)
res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
ast_write(). The reason that this was a bug is that the G.722 frame
originally says it has 320 samples in it (which is correct). However,
when the frame is written to a channel that uses RTP, main/rtp.c modifies
the frame to cut the number of samples in half before it sends it on
the wire. This is to account for the stupid incorrect G.722 spec that
makes it so we have to lie about the number of samples with RTP. I should
probably go and re-work the RTP code so it doesn't modify the frame so
that a bug like this won't happen in the future. However, this change to
MOH is harmless.
main/channel.c:
- I made two fixes in regards to generator timing. Generators use samples
for timing. However, this code assumed 8 kHz samples. In one case, it was
a hard coded 160 samples, that is now written as the sample rate / 50. The
other place was dealing with timing a generator based on frames coming from
the other direction. However, that would have only worked if the sample
rates for the formats in both directions were the same. The code now takes
into account that the sample rates may differ, and scales the generator
samples accordingly.
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r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, 19 Mar 2008) | 12 lines
Merged revisions 110035 via svnmerge from
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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines
Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky
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r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | 25 lines
Merged revisions 109309 via svnmerge from
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r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines
(closes issue #11903)
Reported by: atis
Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.
Sorry for not solving this earlier. I got distracted.
Added the ntest23 regression test, which is mainly a copy of ntest22,
but with a few juicy errors thrown in, to replicate the kind of
error that atis spotted.
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r107791 | tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines
An offhand comment from Russell made me realize that the configuration file
caching would not work properly for users.conf and any other file read from
more than one place. I needed to add the filename which requested the config
file to get it to work properly.
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r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines
Merged revisions 107352 via svnmerge from
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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
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r106399 | juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines
trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense
given our removal of deadagi as an actual application.
(closes issue #12161)
Reported by: explidous
Patches:
res_agi_12161.patch uploaded by juggie (license 24)
Tested by: juggie
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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