https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines
Fix a problem where a 491 response could be sent out of dialog.
This generalizes the fix for issue 13849. The initial fix corrected the
problem that Asterisk would reply with a 491 if a reinvite were received
from an endpoint and we had not yet received an ACK from that endpoint
for the initial INVITE it had sent us. This expansion also allows Asterisk
to appropriately handle an INVITE with authorization credentials if Asterisk
had not received an ACK from the previous transaction in which Asterisk had
responded to an unauthorized INVITE with a 407.
(closes issue #14239)
Reported by: klaus3000
Patches:
14239.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208388 f38db490-d61c-443f-a65b-d21fe96a405b
The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.
(closes issue #15452)
Reported by: alecdavis
Patches:
bug15452.patch uploaded by jpeeler (license 325)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208267 f38db490-d61c-443f-a65b-d21fe96a405b
Pointed out by Matt F.
Also in the case of not using a signaling module, set the law back to the
default as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208155 f38db490-d61c-443f-a65b-d21fe96a405b
This x is one crafty little bugger...
It was used for 2 different things (one of which was only done on PPC) in 1.4.
One of the uses were removed in trunk, and with it went the declaration.
(closes issue #14038)
Reported by: ffloimair
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208113 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines
Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
This change makes URIENCODE and QUOTE behave similarly, since the documentation
states that the argument is not optional, for both.
(closes issue #15439)
Reported by: pkempgen
Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207946 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines
Wait for wink before dialing when using E&M wink signaling
There was already code for other signaling types in dahdi_handle_event to
handle dialing if a dial operation dial string was present. Simply add
SIG_EMWINK to the list.
(closes issue #14434)
Reported by: araasch
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207854 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207680 f38db490-d61c-443f-a65b-d21fe96a405b
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r207573 | jpeeler | 2009-07-20 18:23:18 -0500 (Mon, 20 Jul 2009) | 10 lines
Wait for wink before dialing when using E&M wink signaling
This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up.
(closes issue #14434)
Reported by: araasch
Patches:
emwinkmod uploaded by araasch (license 693)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207599 f38db490-d61c-443f-a65b-d21fe96a405b
This helps to prevent a crash which may occur due to our freeing
garbage due to a struct being uninitialized.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207522 f38db490-d61c-443f-a65b-d21fe96a405b
The registration string can contain an expanded user portion of the
form user@domain. This expanded user portion was stored in
reg->username and parsed each time there is a registration refresh.
Now, the domain portion of the user is parsed and stored separately
in the regdomain field.
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207484 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines
Answer video SDP offers properly when videosupport is not enabled.
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207424 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines
Only do the chan->fdno check in ast_read() in a developer build.
I changed this check to only happen in a dev-mode build. I also added a
comment explaining what is going on. I also made it so that detection of
this situation does not affect ast_read() operation.
(closes issue #14723)
Reported by: seadweller
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207361 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
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r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) | 20 lines
Fixed incoming calls being matched to MSNs without type-of-number prefix added.
For an incoming ISDN call the dialed.number is incorrectly matched against
the configured MSNs in misdn.conf. The numbers passed to the dialplan
include the configured prefix for the dialed.number_type, whereas the
check against the configured MSNs (to decide if the call is accepted at
all), is executed without the configured prefix.
e.g., dialed.number = 241168020, TON = national, configured national
prefix is "0". (This is the TON which is used by ISDN providers in the
Netherlands.)
In chan_misdn.c:cb_events() in case EVENT_SETUP the call to
misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57
lines later the call to read_config() adds the prefix, and the
dialed.number is now 0241168020, which is then used in the dialplan.
misdn_cfg_is_msn_valid() must use the normalized number, too.
JIRA ABE-1912
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207318 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines
channels/chan_misdn.c
channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
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r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
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Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207285 f38db490-d61c-443f-a65b-d21fe96a405b
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r207092 | jpeeler | 2009-07-17 14:13:27 -0500 (Fri, 17 Jul 2009) | 11 lines
Enhance configuration option for overlapdial allowing direction choice
Previously overlap dialing could only be turned on or off for both incoming and
outgoing calls. New parameters incoming, outgoing, and both have been added to
allow further control. There is no change in default behavior with these new
options and allows in band DTMF to be accepted in one direction if required.
(closes issue #14471)
Reported by: eboscani
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207093 f38db490-d61c-443f-a65b-d21fe96a405b
Sig_analog handles allocating the sub channel for callwaiting, so no longer try
to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as
allocated upon success of the alloc_sub callback, which was responsible for the
segfault. Also, the callwaiting and callwaitingcallerid options were being
unconditionally set to true. Now, the options are properly set from
chan_dahdi.conf.
(closes issue #15508)
Reported by: elguero
Tested by: elguero
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206998 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines
SIP incorrect From: header information when callpres is prohib
Some ITSP make use of the "Anonymous" display name to detect a
requirement to withhold caller id across the PSTN. This does
not work if the display name is "Unknown".
(closes issue #14465)
Reported by: Nick_Lewis
Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206939 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines
avoid segfault caused by user error
If the CALLERPRES() dialplan function is set to nothing,
a segfault occurs. This is user error to begin with, but
I'd rather see a cli warning message than have Asterisk
crash on me.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206868 f38db490-d61c-443f-a65b-d21fe96a405b
This readds the proper setting of the flag and is really a continuation of
r205731. The flag was being set properly in sig_analog, but use of the
newly added set_dialing callback allowed for some simplification in
chan_dahdi.
(closes issue #15486)
Reported by: rmudgett
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206767 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines
Merged revision 206700 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
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Fixed chan_misdn crash because mISDNuser library is not thread safe.
With Asterisk the mISDNuser library is driven by two threads concurrently:
1. channels/misdn/isdn_lib.c::manager_event_handler()
2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()
Calls into the library are done concurrently and recursively from
isdn_lib.c.
Both threads can fiddle with the master/child layer3_proc_t lists. One
thread may traverse the list when the other interrupts it and then removes
the list element which the first thread was currently handling. This is
exactly what caused the crash. About 60 calls were needed to a Gigaset
CX475 before it occurred once.
This patch adds locking when calling into the mISDNuser library.
This also fixes some cb_log calls with wrong port parameter.
JIRA ABE-1913
Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206707 f38db490-d61c-443f-a65b-d21fe96a405b
A domain only sip uri <sip:123.123.123.123> would return
123.123.123.123 as callid num. Now, if the username is
missing from a uri, the callerid num field is left empty.
(closes issue #15476)
Reported by: viraptor
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206702 f38db490-d61c-443f-a65b-d21fe96a405b
mmichelson