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r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines
Ensure that the default autoconf CFLAGS are not used.
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
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r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
caller id number empty
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
(closes issue #15839)
Reported by: ebroad
Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel
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r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
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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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r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
sip peer matching by address only with TCP/TLS
This patch removes the contact header matching logic and
adds logic to match all tcp/tls connections by ip only
Review: https://reviewboard.asterisk.org/r/354/
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r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines
make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD
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r215838 | mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
Document that SIPshowpeer and SKINNYshowline now include
the configured parkinglot in their response.
Prodded by snuff-work on #asterisk-dev IRC channel
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r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines
Merged revisions 215682 via svnmerge from
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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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r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
port string to int conversion using sscanf
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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r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
add Parkinglot info to sip show peer <foo> and skinny show line <foo>
If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.
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r215622 | mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
- lock channel before looking for a channel variable
- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
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r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines
like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny lines as well.
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r215466 | dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
SIP support for keep-alive event
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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r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines
Honor configured parkinglot when parking and retrieving parked calls
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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r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines
Merged revisions 215270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
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r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) | 15 lines
Merged revisions 214701 via svnmerge from
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r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines
Modify comment to be a bit more accurate.
We have kept this comment around long enough, that it's pretty clear that we're
keeping the code, because changing the code would require a pretty fundamental
architectural shift. We've also taken criticism in some quarters, because it
was believed that it was referring to the code being nasty. No, the code isn't
nasty, just the operation itself is rather odd. Fixed for eternity (probably
not).
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r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines
Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved.
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
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r214309 | jpeeler | 2009-08-26 18:13:19 -0500 (Wed, 26 Aug 2009) | 12 lines
Add two new dialplan variables when using features
Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
and is set when a dynamic feature is triggered.
(closes issue #14663)
Reported by: tamiel
Patches:
20090313_features.diff uploaded by tamiel (license 712)
Tested by: tamiel
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r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) | 25 lines
Merged revisions 214194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines
ast_write() ignores ast_audiohook_write() results
In ast_write(), if a channel has a list of audiohooks, those
lists are written to and the resulting frame is what ast_write()
should continue with. The problem was the returned audiohook frame
was not being handled at all, and the original frame passed
into it did not contain the mixed audio, so essentially audio
was being lost. One result of this was chan_spy's whisper
mode no longer worked. To complicate the issue, frames
passed into ast_write may either be a single frame, or a list
of frames. So, as the list of frames is processed in the
audiohook_write, the returned frames had to be added to a new
list.
(closes issue #15660)
Reported by: corruptor
Tested by: dvossel
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r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines
Fix storage of greetings when using IMAP_STORAGE
Fix checking if the imapgreetings option is turned on to store the greeting
in IMAP.
(closes issue #14950)
Reported by: noahisaac
Patches:
14950.patch uploaded by mmichelson (license 60)
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