If a device re-INVITEs at the same time as the dialog is hung up, and
if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would
fail to destroy the dialog after a while. This resulted in (most
prominently) file handle leaks.
(Patch reindented by me.)
ASTERISK-20784 #close
ASTERISK-15879 #close
Reported by: Torrey Searle, Nitesh Bansal
Patches:
reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334)
patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418)
Reviewboard: https://reviewboard.asterisk.org/r/4052/
(testcase can be found at r4051)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When starting ice if there is not at least one remote ice candidate with an RTP
component asterisk will crash. This is due to an assertion in pjnath as it
expects at least one candidate with an RTP component. Added a check to make
sure at least one candidate contains an RTP component and at least one candidate
has an RTCP component.
ASTERISK-24383 #close
Review: https://reviewboard.asterisk.org/r/4039/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided
by two can exceed the per-process file limit of 2^20. This patch
ensures the value is capped.
(Patch cleaned up by me.)
ASTERISK-24011 #close
Reported by: Michael Myles
Patches:
safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The underlying library, pjnath, that res_rtp_asterisk uses for ICE
support does not have support for ICE-TCP. As candidates are
passed through directly to it this can cause error messages to occur
when it receives something unexpected (such as a TCP candidate).
This change merely ignores all non-UDP candidates so they never
reach pjnath.
ASTERISK-24326 #close
Reported by: Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If SendMessage encounters an error (such as incorrect input provided to the
action), it will currently return -1. Actions should only return -1 if the
connection to the AMI client should be closed. In this case, SendMessage
causing the client to disconnect is inappropriate.
This patch causes the action to return 0, which simply causes the action to
fail.
Review: https://reviewboard.asterisk.org/r/4024
ASTERISK-24354 #close
Reported by: Peter Katzmann
patches:
sendMessage.patch uploaded by Peter Katzmann (License 5968)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change fixes an issue where ICE candidates put into the SDP did not contain
the 'raddr' and 'rport' information for server reflexive and relay candidates.
#SIPit31
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
'case' works better there.
Originally committed in r375059 and r375060 on 2012-10-16 21:13:08.
ASTERISK-20567 #close
Reported by: Tzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
object ref/unref is attempted. This is similar to the
constructor/destructor lines.
* Fixed refcounter.py to handle skewed objects that have
constructor/destructor states.
* Made refcounter.py highlight the invalid ao2 object refs by putting them
in their own section of the processed output file.
* Made refcounter.py highlight unreffing an object by more than one that
results in a negative ref count and the object being destroyed. The
abnormally destroyed object is reported in the invalid and finalized
object sections of the output.
Review: https://reviewboard.asterisk.org/r/3971/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If faxing fails at a very early stage, then it is possible for
us to pass a NULL t30 state pointer to spandsp, which spandsp
is none too pleased with.
This patch ensures that we pass the correct pointer to spandsp
in the situation where we have not yet set our local t30 state
pointer.
ASTERISK-24301 #close
Reported by Matt Jordan
Patches:
ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV. This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.
The only consumer of this function is manager and it was also changed to use
the return value.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. The number of file descriptors an ioqueue instance can handle is fixed, so we
now spawn the required number to handle the load.
2. Our transport identifiers were exceeding the range supported by pjnath.
3. The TURN client did not set up client binding causing needless bandwidth usage.
4. The code no longer updates address information on each packet.
5. STUN traffic was getting looped back to Asterisk instead of going through the
TURN server.
6. Synchronization now ensures things are completely setup or destroyed.
7. Logging now reflects the target the TURN server is sending to/receiving from
on our behalf.
ASTERISK-23577 #close
Reported by: Jay Jideliov
ASTERISK-23634 #close
Reported by: Roman Skvirsky
Review: https://reviewboard.asterisk.org/r/3982/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge
could cause a bouncing native bridge. In the case of the
dial_LS_options test, this was a remote RTP bridge which caused the
audio path to continually cycle between Asterisk and the remote
endpoints generating a large number of SIP messages and delaying the
test long enough to cause it to fail (checking timing was part of the
test). The root cause was that the code to decide whether to use native
bridging was expecting a time-remaining value of 0 to be the default
instead of the actual default value of -1. A value of 0 or negative
numbers could also be generated by preceding code in some
circumstances. Both issues are addressed in this patch.
ASTERISK-24211 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3987/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_config_text_file_save() currently truncates include files as they
are processed. If a subsequent include file or the main config file has
a permissions error that prevents writing, earlier include files are left
truncated resulting in a frantic search for backups.
This patch causes ast_config_text_file_save to check for write access
on all files before it truncates any of them.
Will be applied 1.8 > trunk.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3986/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The UniMRCP project distributes Asterisk modules that integrate Asterisk with
UniMRCP, and other Asterisk users use the UniMRCP library as well.
Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software
Foundation, is not a compatible license with the GPLv2.
"Please note that this license is not compatible with GPL version 2, because it
has some requirements that are not in that GPL version. These include certain
patent termination and indemnification provisions. The patent termination
provision is a good thing, which is why we recommend the Apache 2.0 license for
substantial programs over other lax permissive licenses."
On the other hand, UniMRCP is a great project and we'd like to let people use
it with Asterisk.
This patch updates the LICENSE text to allow users to link Asterisk with
UniMRCP and distribute the resulting binaries.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The reporter on the issue found some issues when upgrading from version 10 to 11
on 55 hosts.
Two situations that can occur with dynamic registrations.
1. With dnsmgr disabled, if the host is not resolvable we are not trying to
resolve the host again when it is time to attempt to register again. This
results in never registering to the host.
2. With dnsmgr enabled, when the host is temporarily not resolvable the
address is set to 0.0.0.0:0 and then when the host is resolvable the port
is not being restored and stays set to 0.
This patch resolves these two issues by:
* Storing the hostname so that it can be used for resolving with DNS.
* Resolve the hostname on the next scheduled attempt to register.
* Storing the port used to reach the host so that when the hostname is
resolvable again, we can set the port again if the port is still unset after
looking up the host.
ASTERISK-23767 #close
Reported by: David Herselman
Tested by: David Herselman, Michael L. Young
Patches:
asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3856/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.
ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
plus1.diff submitted by Badalian Vyacheslav (license 5249)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to a faulty function for debugging reference decrementing, it was possible
to reduce the refcount on the wrong object if two moh classes of the same name
were in the moh class container.
(closes issue ASTERISK-22252)
Reported by: Walter Doekes
Patches:
18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a user does not provide a port in the fromdomain setting, chan_sip will set
the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
then get used unilaterally in certain places. This causes issues with TLS,
where the default port is expected to be 5061.
This patch modifies chan_sip such that fromdomainport is only used if it is
not the standard SIP port; otherwise, the port from the SIP pvt's recorded
self IP address is used.
Review: https://reviewboard.asterisk.org/r/3893/
ASTERISK-24178 #close
Reported by: Elazar Broad
patches:
fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR. This does nothing but
clutter up the logs with messages that may be perfectly acceptable. Just
because a variable wasn't in the context doesn't mean it's an error. Maybei
t's optional or just needs to be defaulted or ignored.
This patch changes the log level from ERROR to DEBUG. If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level
as needed.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.
ASTERISK-24225 #close
Reported by: dimitripietro
Tested by: dimitripietro
patches:
jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.
Review: https://reviewboard.asterisk.org/r/3912/
ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
1.8.diff uploaded by cloos (License 5956)
10.diff uploaded by cloos (License 5956)
11.diff uploaded by cloos (License 5956)
12.diff uploaded by cloos (License 5956)
13.diff uploaded by cloos (License 5956)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.
Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
