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r255592 | tilghman | 2010-03-31 14:13:02 -0500 (Wed, 31 Mar 2010) | 22 lines
Recorded merge of revisions 255591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines
Ensure line terminators in email are consistent.
Fixes an issue with certain Mail Transport Agents, where attachments are not
interpreted correctly.
(closes issue #16557)
Reported by: jcovert
Patches:
20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: ebroad, zktech
Reviewboard: https://reviewboard.asterisk.org/r/544/
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r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines
Update confusing documentation for tlsbindaddr.
Update some confusing documentation for the tlsbindaddr
option in sip.conf.sample. Point at a link instead which
has better documentation.
(closes issue #17054)
Reported by: klaus3000
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r254454 | mmichelson | 2010-03-25 11:04:48 -0500 (Thu, 25 Mar 2010) | 50 lines
Recorded merge of revisions 254452 via svnmerge from
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r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines
Several fixes regarding RFC2833 DTMF detection.
Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes:
Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1
seqno 4: DTMF 1
seqno 6: DTMF 1 (end)
seqno 5: DTMF 1
seqno 7: DTMF 1 (end)
seqno 8: DTMF 1 (end)
Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too:
seqno 9: DTMF 1
seqno 10: DTMF 1 (end)
seqno 11: DTMF 1 (end)
seqno 13: DTMF 2
seqno 12: DTMF 1 (end)
seqno 14: DTMF 2
seqno 15: DTMF 2 (end)
seqno 16: DTMF 2 (end)
seqno 17: DTMF 2 (end)
In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF.
Fix 2. The second change in place is to fix an issue like the following:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1 (end) *packet lost*
seqno 4: DTMF 1 (end) *packet lost*
seqno 5: DTMF 1 (end) *packet lost*
seqno 6: DTMF 2
When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list.
Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem
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r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) | 78 lines
Merged revisions 254235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines
Ensure that monitor recordings are written to the correct location (again)
This is an extension to 248860. As such the dialplan test has been extended:
; non absolute path, not combined
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001)
; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2)
exten => 5041, n, dial(sip/5001)
; no path, not combined
exten => 5042, 1, monitor(wav,monitor_test3)
exten => 5042, n, dial(sip/5001)
; combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5)
exten => 5043, n, dial(sip/5001)
; combined: changemonitor from no path to non absolute path
exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before
exten => 5044, n, dial(sip/5001)
; non absolute path, combined
exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m)
exten => 5045, n, dial(sip/5001)
; absolute path, combined
exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m)
exten => 5046, n, dial(sip/5001)
; no path, combined
exten => 5047, 1, monitor(wav,monitor_test10,m)
exten => 5047, n, dial(sip/5001)
; combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m)
exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12)
exten => 5048, n, dial(sip/5001)
; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m)
exten => 5049, n, changemonitor(tmp/jeff/monitor_test14)
exten => 5049, n, dial(sip/5001)
; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m)
exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16)
exten => 5050, n, dial(sip/5001)
; combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m)
exten => 5051, n, changemonitor(monitor_test18)
exten => 5051, n, dial(sip/5001)
; not combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20)
exten => 5052, n, dial(sip/5001)
; not combined: changemonitor from no path to non absolute
exten => 5053, 1, monitor(wav,monitor_test21)
exten => 5053, n, changemonitor(tmp/jeff/monitor_test22)
exten => 5053, n, dial(sip/5001)
; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23)
exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24)
exten => 5054, n, dial(sip/5001)
; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24)
exten => 5055, n, changemonitor(tmp/jeff/monitor_test25)
exten => 5055, n, dial(sip/5001)
; not combined: changemonitor from no path to absolute
exten => 5056, 1, monitor(wav,monitor_test26)
exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27)
exten => 5056, n, dial(sip/5001)
; not combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28)
exten => 5057, n, changemonitor(monitor_test29)
exten => 5057, n, dial(sip/5001)
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r254050 | jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
Exit native bridging early for greater timing accuracy with warnings
This changes native bridging to break one millisecond early so that the more
accurate timeval calculations done in the generic bridge can be performed using
the bridge config. Currently the time between exiting native bridging slightly
late can sometimes cause a large enough discrepancy for warnings to be missed.
For the record, 1.4 does not attempt to native bridge at all when warnings are
enabled.
(closes issue #15815)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/577/
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r253490 | alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19 lines
prevent segfault if bad magic number is encountered.
internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but
internal_ao2_ref continues on, causing segfault.
Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is
called, A02_MAGIC is being destroyed (or a wrong pointer) by the time
internal_ao2_ref uses INTERNAL_OBJ.
internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number.
(issue #17037)
Reported by: alecdavis
Patches:
bug17037.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) | 13 lines
Merged revisions 252766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) | 6 lines
Don't treat warnings as errors for muted.
muted supports OS X, but uses functions marked as deprecated in 10.6. However,
the functions are still supported, so just ignore the warnings for now and
allow the build to proceed.
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r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines
Merged revisions 252761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines
Additional extensions.ael global variable fixes.
Fixing up a couple more overlapping global variable namespaces shared with
extensions.conf.sample. Also noticed a few of the lines that were commented
out didn't have the closing semi-colon so I added that as well.
(issue #17035)
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r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines
Merged revisions 252533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines
Update extensions.ael file to not overlap extensions.conf.
Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
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r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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