We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
(closes issue #12471)
Reported by: mthomasslo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
(closes AST-152)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Closes AST-158
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: john8675309
Patches:
patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309
This patch causes the setcid func to update the CDR
clid after setting the channel field.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was initiated but not yet completed. If the bridge is immediately retried,
then we may send a storm of TXREQ packets, even though the first set is sent
reliably (retransmitted). Fixes AST-137.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
then that new entry will not be processed promptly, but must wait for either a
future entry to start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch expire, then
the new entries must wait until another new entry is added to be processed.
This was a rather weird race condition, really. Fixes AST-147.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"I guess that having only ip-phones in mind is not a good approach. Since it is
possible to have a sip proxy connected to asterisk we could receive a 407
(unauthorized) or 483 (too many hops) as response and dialog ending would not be
a good behavior."
So modified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fails, and the resulting integer is garbage. Thus, we must initialize the
integer and check it afterwards for success.
(closes issue #14000)
Reported by: folke
Patches:
asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
add channels/busy.h and channels/ringtone.h to the repository instead of generating them repeatedtly; most users do not change the settings to build them, but the Makefile rules are still there if they wish to do so
ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: tzafrir
When compiling against Zaptel dahdi_compat will now only define all the DAHDI defines if the Zaptel define is present. Also, there is no such thing as DAHDI_PRI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
merging the fix into 1.6.0 and realized I missed
the code in the h-exten block, and didn't catch it
because my test case had the h-exten commented out.
So, this corrects the code I missed, as a
preventative against another crash report.
Tested with the h-exten defined, all is well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: mdu113
This one is totally my fault. The code doesn't even
create a bridge if the channel CDR has POST_DISABLED.
I didn't check for that at the end of the bridge.
Fixed with a few small insertions. Tested. Looks
good. No cdr generated, no crash, no unnecc. data
objects created either.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
frame in a freed ast_filestream. This patch makes use of the
ao2 functions to make sure that we do not free an ast_filestream
structure until the embedded ast_frame has been "freed" as well.
(closes issue #13496)
Reported by: fst-onge
Patches:
filestream_frame_1_4.diff uploaded by putnopvut (license 60)
Tested by: putnopvut
Closes AST-89
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
