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reload.
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counter roof to something higher.
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unset minimum = -1)
Patch by oej
closes issue #15621
Reported by: fnordian
Tested by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch removes the contact header matching logic and
adds logic to match all tcp/tls connections by ip only.
Thanks to oej for finding the issue and suggesting solutions.
Review: https://reviewboard.asterisk.org/r/354/
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Suggested on the -dev list, and implemented in an alternate way by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines
make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009) | 2 lines
make chan_sip compile under devmode again
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r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009) | 4 lines
Make sure 'start' is always initialized.
This is the same as rev 216222 in trunk but 1.4 is affected as well
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There was a problem in the function responsible for doing peer matching by
IP address and port number such that during the second pass for checking for
a peer configured with insecure=port, it would end up treating every peer as
if it had been configured that way. These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking properly.
This problem was introduced when SIP peers were converted to astobj2. Many
thanks to dvossel for noticing this while working on another peer matching
issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up
one of these is often a good way to get into Asterisk development and getting a lot of developers in
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.
If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.
Thank you for your help.
</end of long commit message>
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r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines
Merged revisions 216262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines
Add a plain text version of the IAX2 security document.
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Makes asterisk compile with --enable-dev-mode
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Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines
Add a note about IAX2 to UPGRADE.txt.
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r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
Add IAX2 security document related to AST-2009-006.
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This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.
(closes issue #15771)
Reported by: jtodd
Patches:
language-criteria.txt uploaded by jtodd
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r216000 | dvossel | 2009-09-03 13:32:32 -0500 (Thu, 03 Sep 2009) | 7 lines
Merge code associated with AST-2009-006
(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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(closes issue #14573)
Reported by: pj
Patches:
sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
modified by oej
Tested by: pj
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the configured parkinglot in their response.
Prodded by snuff-work on #asterisk-dev IRC channel
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(closes issue #15764)
Reported by: elguero
Change-type: bugfix
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This did not function in the way that was intended, causing more compatibility
issues than it solved. It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines
Re-send non-100 provisional responses to prevent cancellation
From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
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ISDN PTMP CPE spans.
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There are several instances where a port is parsed
from a uri or some other source and converted to
an int value using atoi(), if for some reason the
port string is empty, then a standard port is used.
This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().
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If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.
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- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
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made analog_set_linear_mode return back the mode that was being overwritten
so it could be restored later.
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on OS X).
Also, a Makefile fix for Darwin (OS X).
(closes issue #14542)
Reported by: jtodd
Patches:
20090901__issue14542.diff.txt uploaded by tilghman (license 14)
Tested by: jtodd, tilghman
Change-type: bugfix
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Now, the scheme passed to parse_uri can either be a
single scheme, or a list of schemes ',' delimited.
This gets rid of the whole problem of having to create
two buffers and calling parse_uri twice to check for
separate schemes.
Review: https://reviewboard.asterisk.org/r/343/
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a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny lines as well.
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keep-alive events are used by Sipura/Linksys for NAT keepalive.
There currently don't appear to be any problems with NAT, but
everytime a keep-alive event is received, Asterisk responds with a
"489 Bad event". This error may indicate to a user that NAT
problems exist just because this even is not supported. Now,
rather than respond with an error, the packet is consumed and
a "200 ok" is sent just to indicate we received the packet.
(issue #15084)
Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
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Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer
into the newly created channel.
(closes issue #15538)
Reported by: gracedman
Patches:
2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7)
With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak
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