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being passed to printf()-like functions and ignored results from read()/write() and friends
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@153337 f38db490-d61c-443f-a65b-d21fe96a405b
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there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
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(closes issue #13215)
Reported by: qwell
Patches:
2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak
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on it.
(closes issue #12669)
Reported by: sbisker
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7962/7965, 7975.
Thanks to Greg Oliver for providing me the required information.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@113454 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #9077)
Reported by: pj
Patches:
earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn
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when dialing a channel that does not provide progress (analog ZAP lines)
The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items
(closes issue #12239)
Reported by: DEA
Patches:
chan_skinny-call-log.txt uploaded by DEA (license 3)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@109171 f38db490-d61c-443f-a65b-d21fe96a405b
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on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
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Issue 11586, patch by sobomax.
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Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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(ASA-2007-019)
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Also make note of the "A" option of date format.
Issue 9779, modifications by DEA, wedhorn, and myself.
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If this happens during call setup, we need to give up.
Issue 10325.
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Also rearrange messages to better fit with what a wireshark trace shows it should be.
Issue 10299, initial patch and solution by sbisker, modified by me to fit with wireshark trace.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77947 f38db490-d61c-443f-a65b-d21fe96a405b
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"From: 5555" rather than "From: s"
Issue 10358, initial patch by DEA, reworked by me to use S_OR, tested by sbisker
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Issue 10291, patch by DEA.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77890 f38db490-d61c-443f-a65b-d21fe96a405b
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the 7921),
due to sending packets in the wrong order during hangup.
Also make sure we clear tones/messages on the correct line/instance.
Issue 10291, patch by DEA, tested by sbisker and myself.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77883 f38db490-d61c-443f-a65b-d21fe96a405b
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Don't try to play dialtone in some unimplemented features.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@76656 f38db490-d61c-443f-a65b-d21fe96a405b
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Issue 10276.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@76620 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: mvanbaak
Patches:
chan_skinny_openbsd.diff uploaded by mvanbaak (license 7)
Add another OS that has to use the Macros for byte ordering.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@76519 f38db490-d61c-443f-a65b-d21fe96a405b
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Cancel ringing if endpoint hangs up before answering.
Fixes were backported from trunk (there was apparently a bit of confusion during merge of a previous patch).
(closes issue #10241)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@76178 f38db490-d61c-443f-a65b-d21fe96a405b
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immediately after/during sending a digit.
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recent ast_ time stuff from Tilghman
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Issue 9245, patch by slimey.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@75711 f38db490-d61c-443f-a65b-d21fe96a405b
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Issue 9887, patches by DEA
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 Jul 2007) | 3 lines
Properly check for the length in the skinny packet to prevent an invalid memcpy.
(ASA-2007-016)
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actually do much of anything.
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Issue 10136, fix provided by DEA.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@74839 f38db490-d61c-443f-a65b-d21fe96a405b
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reported by gzero)
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Issue 9958.
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correct length.
Issue 9887, patch by DEA.
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instead of just at registration.
Issue 9152, yet another patch by DEA.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@67421 f38db490-d61c-443f-a65b-d21fe96a405b
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If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device.
Issue 8394, patch by DEA.
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This patch makes chan_skinny usable again. I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was
written by DEA)
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refinement, but this won't have as many folks bothered.
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(issue #9002)
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
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Issue 8995, initial patch and research done by wedhorn
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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while in use will not result in crashes. (issue #8897 reported by junky)
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The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines
ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly
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