Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session).
(closes issue #16980)
Reported by: wedhorn
Patches:
skinny-clean06b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, DEA
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines
Merged revisions 280306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines
Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229
Review: https://reviewboard.asterisk.org/r/813/
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Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699.
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r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines
Fix inband DTMF detection on outgoing ISDN calls.
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines
Fix parsing error in sip_sipredirect().
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines
Merged revisions 279784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
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r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines
transaction matching using top most Via header
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
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r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "dahdi show channels" extension column previously only showed the
called number of an incoming call. It now shows the called number for an
outgoing call as well.
(closes issue #17653)
Reported by: amazinzay
Patches:
issue17653_trunk.txt uploaded by rmudgett (license 664)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This initially was created to work around the issue of
using a string comparison instead of a binary comparison
for IP addresses. It evolved a bit when test cases were
created and it was discovered that comparison of URI
parameters was not working exactly as it should.
sip_uri_cmp() and its helpers have been moved to reqresp_parser.c
and a new test has been added.
(closes issue #17662)
Reported by: oej
Review: https://reviewboard.asterisk.org/r/792
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
FXS lines normally connect to a telephone. However, when FXS lines are routed
to an external PBX or Key System to act as "external" or "CO" lines, it is
extremely difficult, if not impossible for the external PBX to know when
the call has been disconnected without receiving a polarity reversal on the line.
Now using answeronpolarityswitch and hanguponpolarityswitch keywords that
previously were used only for FXO ports, now applies like functionality for
an FXS port, but from the connected equipment's point of view.
(closes issue #17318)
Reported by: armeniki
Patches:
fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/797/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "dahdi show channels" CLI command still reports the DNID of the
previous call even if the call is already hang up. The "dahdi show
channels" command of older releases clear the DNID once the channel is
hang up.
Regression from the sig_analog/sig_pri extraction from chan_dahdi.
(closes issue #17623)
Reported by: klaus3000
Patches:
issue17623.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
[RFC3265]3.1.6.1....
The notifier MAY also check that the duration in the "Expires" header
is not too small. If and only if the expiration interval is greater
than zero AND smaller than one hour AND less than a notifier-
configured minimum, the notifier MAY return a "423 Interval too
small" error which contains a "Min-Expires" header field. The "Min-
Expires" header field is described in SIP [1].
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes some cases of no outgoing calls on FXO before an incoming call.
Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO
(KS/GS) channels.In some cases the bit would not be initialized properly
before the first inbound call and thus prevent an outgoing call.
If those tests are actually required by anybody, they should define
DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c .
(closes issue #14577)
Reported by: jkroon
Patches:
asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by frawd (license 610)
Tested by: frawd
Review: https://reviewboard.asterisk.org/r/699/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If Asterisk sends a 4xx error and the other side sends a CANCEl
before receiving the 4xx and responding with the ACK, Asterisk
will process the CANCEL and send a 487 Request Terminated as
a new final response to the INVITE. Since we are issuing a new
final response to the INVITE, the old one must be pretend_acked
else it will keep retransmitting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
(closes issue #17665)
Reported by: mmichelson
Patches:
17665.diff#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The issue here is that passing an array to a function prohibits the ARRAY_LEN
macro from returning the real size. To avoid this the size is now defined and
use of ARRAY_LEN is avoided.
(closes issue #15718)
Reported by: alecdavis
Patches:
bug15718.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
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