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to make merging easier. These changes are already on trunk.
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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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r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines
Merged revisions 207155 via svnmerge from
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r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines
Fix format specifier to print out an unsigned long long.
Yep, it's even ifdefed out code. But it made it to the RR list...
(closes issue #14726)
Reported by: lmadsen
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r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines
Update some missing allowed options for overlapdial
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r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
sip option flags handled incorrectly
(closes issue #15376)
Reported by: Takehiko Ooshima
Tested by: dvossel, Takehiko_Ooshima
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r206998 | jpeeler | 2009-07-17 12:02:44 -0500 (Fri, 17 Jul 2009) | 14 lines
Fix segfault in sig_analog when using callwaiting, respect callwaiting options
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
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r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines
Merged revisions 206938 via svnmerge from
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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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r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines
TIMEOUT(absolute) returned negative value.
(closes issue #15513)
Reported by: ys
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r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines
Merged revisions 206872 via svnmerge from
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r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines
error in iax.conf related IP-based access control
(closes issue #15518)
Reported by: pkempgen
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r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines
Merged revisions 206867 via svnmerge from
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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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r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines
Merged revisions 206807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines
Fix a memory leak.
(closes issue #15517)
Reported by: adomjan
Patches:
func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487)
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r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
Session timer were not activated if Supported header field in INVITE had both "timer" and other options.
(closes issue #15403)
Reported by: makoto
Patches:
sip-session-timer.patch uploaded by makoto (license
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r206767 | jpeeler | 2009-07-15 17:02:55 -0500 (Wed, 15 Jul 2009) | 10 lines
The dialing flag was mistakingly removed from sig_pri.
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
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r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines
Merged revisions 206706 via svnmerge from
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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r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
callerid(num) is wrong when username is missing
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
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r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines
Merged revisions 206635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line
Only print debug info in codec_dahdi if we are asking for it.
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r206566 | jpeeler | 2009-07-14 15:01:10 -0500 (Tue, 14 Jul 2009) | 8 lines
Restore some missing functionality to sig_analog.
The main purpose of this commit is to restore missing functionality present in
the ss_thread before all the sig related work was done. Two of the biggest
missing things were distinctive ring detection and cid handling for V23.
fxsoffhookstate and associated mwi variables have been moved inside sig_analog
as they were not being set properly as well.
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r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines
Document all meetme realtime fields, and in the process, make some field lengths more consistent.
(closes issue #15493)
Reported by: lasko
Patches:
meetme.diff uploaded by lasko (license 833)
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r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines
Merged revisions 206487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines
Fixes several call transfer issues with chan_misdn.
* issue #14355 - Crash if attempt to transfer a call to an application.
Masquerade the other pair of the four asterisk channels involved in the
two calls. The held call already must be a bridged call (not an
applicaton) or it would have been rejected.
* issue #14692 - Held calls are not automatically cleared after transfer.
Allow the core to initate disconnect of held calls to the ISDN port. This
also fixes a similar case where the party on hold hangs up before being
transferred or taken off hold.
* JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
Do not simply block passing the hangup event on held calls to asterisk
core.
* Fixed to allow held calls to be transferred to ringing calls.
Previously, held calls could only be transferred to connected calls.
* Eliminated unused call states to simplify hangup code.
* Eliminated most uses of "holded" because it is not a word.
(closes issue #14355)
(closes issue #14692)
Reported by: sodom
Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
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r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines
Merged revisions 206385 via svnmerge from
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r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines
Merged revisions 206384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
Ensure apathetic replies are sent out on the proper socket.
chan_iax2 supports multiple address bindings. The send_apathetic_reply()
function did not attempt to send its response on the same socket that the
incoming message came in on.
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r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines
Merged revisions 206284 via svnmerge from
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r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines
Fix some memory leaks in chan_misdn.
JIRA ABE-1911
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r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
SIP register not using peer's outbound proxy
If callbackextension is defined for a peer it successfully causes
a registration to occur, but the registration ignores the
outboundproxy settings for the peer. This patch allows the
peer to be passed to obproxy_get() in transmit_register().
(closes issue #14344)
Reported by: Nick_Lewis
Patches:
callbackextension_peer_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/294/
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r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
Update comments about the level of T.38 support in Asterisk.
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r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines
Merged revisions 205877 via svnmerge from
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r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines
Merged revisions 205776 via svnmerge from
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r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
Merged revisions 205775 via svnmerge from
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r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
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r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines
Merged revisions 205804 via svnmerge from
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r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines
SIP registration auth loop caused by stale nonce
If an endpoint sends two registration requests in a very short
period of time with the same nonce, both receive 401 responses
from Asterisk, each with a different nonce (the second 401
containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match
the current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if
the endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a
newly generated nonce... This loop goes on and on.
There appears to be a simple fix for this. If the nonce from
the request does not match our nonce, but is a good response
to a previous nonce, instead of sending a 401 with a newly
generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no
nonce can be responded to twice though.
(closes issue #15102)
Reported by: Jamuel
Patches:
patch-bug_0015102 uploaded by Jamuel (license 809)
nonce_sip.diff uploaded by dvossel (license 671)
Tested by: Jamuel
Review: https://reviewboard.asterisk.org/r/289/
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r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
Merged revisions 205775 via svnmerge from
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r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
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r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines
Fix some remaining T.38 negotiation problems in app_fax.
Revision 205696 did not quite fix all the issues with the T.38 negotiation
changes and app_fax; this patch corrects them, along with a couple of other
minor issues.
(closes issue #15480)
Reported by: dimas
Patches:
test2-15480.patch uploaded by dimas (license 88)
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r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
(i.e. When libpri generates the event PRI_EVENT_ANSWER.)
(closes issue #15420)
Reported by: scottbmilne
Patches:
bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis
(closes issue #15416)
Reported by: avinoash
(closes issue #15389)
Reported by: alecdavis
This patch should also fix the following issue:
(issue #15205)
Reported by: vinsik
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
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r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines
Merged revisions 205599 via svnmerge from
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r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines
Changing ast_samp2tv to not use floating point.
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r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines
Merged revisions 205471 via svnmerge from
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r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This
is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas
that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/
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r205562 | mvanbaak | 2009-07-09 16:10:01 +0200 (Thu, 09 Jul 2009) | 2 lines
make this compile again under devmode
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r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
pthread_self returns a pthread_t which is not an unsigned int on all
pthread implementations. Casting it to an unsigned int fixes compiler warnings.
Tested on OpenBSD and Linux both 32 and 64 bit
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r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines
Merged revisions 205409 via svnmerge from
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r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change
fixes a compile time error with chan_vpb as well.
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r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines
Merged revisions 205349 via svnmerge from
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r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
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r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines
Merged revisions 205288 via svnmerge from
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r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line
Update config.guess and config.sub from the savannah.gnu.org git repo.
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r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines
Merged revisions 205215 via svnmerge from
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r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines
ast_samp2tv needs floating point for 16khz audio
In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
The .5 is currently stripped off because we don't calculate
using floating points. This causes madness with 16khz audio.
(issue ABE-1899)
Review: https://reviewboard.asterisk.org/r/305/
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r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines
Merged revisions 205188 via svnmerge from
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r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines
Add redirection warnings for the invalid language codes previously removed.
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r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
Use tabs instead of spaces for indentation.
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r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
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This patch adds reference counting for sip dialogs into 1.6.0.
When proc_session_timer() is called from the scheduler thread
it has no guarantee the session timer's dialog won't be freed
from underneath it. Now the session timer holds a reference
to the dialog, preventing it from being destroyed during the
middle of proc_session_timer().
(closes issue #13623)
Reported by: Nik Soggia
Review: https://reviewboard.asterisk.org/r/302/
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r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines
Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels.
This change allows applications that request T.38 negotiation on a channel that
does not support it to get the proper indication that it is not supported, rather
than thinking that negotiation was started when it was not.
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r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines
Merged revisions 204834 via svnmerge from
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r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
Removed confusing warning message "Got Busy in Connected State"
If an incoming mISDN call is answered with the Answer application and a
subsequent Dial gets a busy endpoint then it is valid for that already
connected channel to get the busy indication. Asterisk will play the busy
tones until the dialplan plays something else or hangs up the call.
(closes issue #11974)
Reported by: fvdb
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r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
Merged revisions 204681 via svnmerge from
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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dialog_unref() and dialog_ref() in 1.6.0 where only place holders
for reference counting once it was implemented. The functions
did nothing but return the pointer on ref and NULL on unref. These
calls have been removed to make way for a patch that actually does
dialog ref counting.
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r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines
Merged revisions 204556 via svnmerge from
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r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022)
Reported by: greenfieldtech
Patches:
20090519__issue15022.diff.txt uploaded by tilghman (license 14)
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r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines
Merged revisions 204474 via svnmerge from
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r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line
Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing.
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r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines
Recorded merge of revisions 204469 via svnmerge from
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r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346)
Reported by: volivier
Patches:
20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: volivier
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