Age | Commit message (Collapse) | Author |
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RFC 6234 is an update to RFC 3174 from which the code was originally taken.
It has a slightly better code, and a better phrased license (simple 3-clause
BSD).
* main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
* include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234.
* Removed unused include of asterisk/sha1.h from main/channels.c
Review: https://reviewboard.asterisk.org/r/1503/
Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263
Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280
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* Fixed race between calling an AMI action callback and unregistering that
action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
* Fixed potential memory leak if an AMI action failed to get registered
because is already was registered. Part of the ao2 conversion.
* Fixed AMI ListCommands action not walking the actions list with a lock
held.
* Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage.
* Fix AMI Originate action Variable header requiring a space after the
header colon. Reported by Yaroslav Panych on the asterisk-dev list.
* Increased the number of listed variables allowed per AMI Originate
action Variable header to 64.
* Fixed AMI GetConfigJSON action output format.
* Fixed usage of res contents outside of scope in append_channel_vars().
* Fixed inconsistency of config file channelvars option. The values no
longer accumulate with every channelvars option in the config file. Only
the last value is kept to be consistent with the CLI "manager show
settings" command.
(closes issue ASTERISK-18479)
Reported by: Jaco Kroon
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Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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(closes issue AST-654)
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r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) | 8 lines
On astdb conversion, also warn about permissions requirements
The user running Asterisk must have permission to the directory
the Asterisk database resides in since SQLite 3 needs to be able
to create a journal file.
(closes issue ASTERISK-18174)
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r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines
Add astdb conversion utility for Berkeley to SQLite 3
If someone wants to backtrack from Asterisk 1.8 to 10 they can use the
astdb2bdb utility to convert the database back to the Berkeley format
that Asterisk 1.8 uses.
Review: https://reviewboard.asterisk.org/r/1502/
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r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
Add a missing file for the astdb2bdb conversion utility
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r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines
Merged revisions 340164 via svnmerge from
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r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines
Updated chan_sip to place calls on hold if SDP address in INVITE is ANY
This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::.
In this case, the call should be placed on hold. Previously, we checked for
the address being null; this patch keeps that behavior but also checks for
the ANY IP addresses.
Review: https://reviewboard.asterisk.org/r/1504/
(closes issue ASTERISK-18086)
Reported by: James Bottomley
Tested by: Matt Jordan
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r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines
Merged revisions 340108 via svnmerge from
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r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines
Load the proper XML documentation when multiple modules document the same application.
This patch adds an optional "module" attribute to the XML documentation spec
that allows the documentation processor to match apps with identical names from
different modules to their documentation. This patch also fixes a number of
bugs with the documentation processor and should make it a little more
efficient. Support for multiple languages has also been properly implemented.
ASTERISK-18130
Review: https://reviewboard.asterisk.org/r/1485/
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Added some data to skinny packet structures to make compatible
with v17. Added protocolversion to device, set on registration
based on the version provided by device.
v17 includes some increased ip space for ip6. This patch increases
ip space in the packets but still only uses ip4. Some packet
structures duplicated (ip4 and ip6 types). ip4 type used unless
version is greater or equal to 17.
Tested by snuff and myself on 7961 with recent 8.5 firmware. Also
tested compatible with old 7960 and older 30VIPs.
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Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages
in v17 that are greater than the old 1000 bytes. Also add some
useful logging regarding packet and session handling.
A device (with protocol v17) was sending a packet with length
greater than 1000 which resulted in the TCP session being
destroyed and registration being retryed.
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r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines
Return -1 to skinny_session if register rejected.
If device registration is rejected, return -1 so that the session is
destroyed immediately. Previously, a segfault would occur on a
graceful shutdown if a register is rejected and the skinny_session
has not yet timed out.
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r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines
Remove log message on traverse session list.
On destroying a session, a list of sessions is traversed to find the
matching session. For each session not matching, skinny erroneously
logged that the session was not matched. While technically correct
the message was misleading, and tended to indicate errors that
were not there.
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r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines
Merged revisions 339938 via svnmerge from
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r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines
Fix compilation issue, caused by missed session structure
(closes issue ASTERISK-18694)
Reported by: alex70
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r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines
Merged revisions 339884 via svnmerge from
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r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines
Fix segfault in Unistim channel
(closes issue ASTERISK-18638)
Reported by: jonnt
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r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines
Merged revisions 339830 via svnmerge from
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r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines
Fix char array cast as short array in send_client() function (for ARM
platform)
(closes issue ASTERISK-17314)
Reported by: jjoshua
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r339777 | rmudgett | 2011-10-07 14:36:24 -0500 (Fri, 07 Oct 2011) | 12 lines
Merged revisions 339776 via svnmerge from
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r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines
Initialize option flags for SendURL application.
(closes issue ASTERISK-18574)
Reported by: marcelloceschia
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r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011) | 10 lines
Fixed segfault on core stop gracefully.
There was an issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same ast_format_cap. On
destroying, a segfault occured on the second call to the same struct.
skinny reload now works again as well.
Tested by snuff (in trunk) and myself.
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r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines
Merged revisions 339719 via svnmerge from
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r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
Fix regression in configure script for libpri capability checks.
JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
2 persistence issues with some telcos. ASTERISK-18535 attempted to fix
the unexpected requirement that libpri *must* have that feature to work
with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
optional features required. Unfortunately, I thought
AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
deleted those lines for libpri. The result was the HAVE_PRI_xxx defines
that control the ability to use optional libpri features were also
deleted.
* Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage of if the
code supports the feature.
(closes issue ASTERISK-18687)
Reported by: Norbert
Tested by: rmudgett
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There was an issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same ast_format_cap. On
destroying, a segfault occured on the second call to the same struct.
skinny reload now works again as well.
Tested by snuff and myself.
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r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines
Merged revisions 339625 via svnmerge from
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r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines
Fix debugging messages generated by 'udptl debug'.
* Makes chan_sip set the tag to the channel name.
* Fixes received debug message sequence number.
* Removed tx/rx debug message type since it was hard coded to 0.
* Made udptl.c logged message header consistent if possible: "UDPTL (%s): ".
* Removed unused rx_expected_seq_no from struct ast_udptl.
(closes issue ASTERISK-18401)
Reported by: Kevin P. Fleming
Patches:
jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Matthew Nicholson
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r339586 | lmadsen | 2011-10-06 08:43:21 -0500 (Thu, 06 Oct 2011) | 16 lines
Merged revisions 339566 via svnmerge from
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r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines
Update prep_tarball script to download pre-exported documentation.
I've updated the prep_tarball script to now download the pre-exported documentation
from the Asterisk wiki. This will give us more control over what is being included
in the tarball releases, and will make both the PDF and HTML exported documentation
look much better (especially when viewing from a console).
(Closes issue ASTERISK-18677)
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r339512 | rmudgett | 2011-10-05 12:01:46 -0500 (Wed, 05 Oct 2011) | 9 lines
Merged revisions 339511 via svnmerge from
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r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line
Fix Dial F option notes formatting.
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r339508 | rmudgett | 2011-10-05 11:35:02 -0500 (Wed, 05 Oct 2011) | 18 lines
Merged revisions 339504,339506 via svnmerge from
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r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines
Add missing documentation of required AMI action Challenge AuthType header.
(closes issue ASTERISK-18554)
Reported by: Vlad Povorozniuc
Patches:
__20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen
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r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line
Fix XML error in AMI action Challenge.
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r339507 | mnicholson | 2011-10-05 11:32:59 -0500 (Wed, 05 Oct 2011) | 10 lines
Merged revisions 339505 via svnmerge from
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r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines
The app name in the documentation must match what we register the application
as.
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Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no
to enable dialplan faxdetect allowing more flexibility.
as soon as a fax tone is detected the framehook is removed.
there is a penalty involved in running this framehook on
non G711 channels as they will be transcoded.
CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice.
(Closes issue ASTERISK-18569)
Reported by: Myself
Reviewed by: Matthew Nicholson, Kevin Fleming
Review: https://reviewboard.asterisk.org/r/1116/
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r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines
Only change the capabilities on the gateway when
the session is been destroyed there is still
a race condition that ends in a segfault.
if the caps are changed the logic in res_fax_spandsp
will run T30 code not gateway code to end the session.
this has been experienced on a "slower" under spec system.
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r339407 | rmudgett | 2011-10-04 17:56:25 -0500 (Tue, 04 Oct 2011) | 15 lines
Merged revisions 339406 via svnmerge from
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r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines
Make always create the MOH directory (/var/lib/asterisk/moh).
(closes issue ASTERISK-18409)
Reported by: abelbeck
Patches:
asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck
Tested by: abelbeck, Michael Keuter
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r339353 | jrose | 2011-10-04 14:44:02 -0500 (Tue, 04 Oct 2011) | 18 lines
Merged revisions 339352 via svnmerge from
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r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines
Removes improper use of sound 'and' in German language mode from application saynumber
Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs
und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk
will only say the 'and' word between the single digit and double digit places.
(closes issue ASTERISK-18212)
Reported By: Lionel Elie Mamane
Patches:
upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane
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r339298 | jrose | 2011-10-04 09:09:50 -0500 (Tue, 04 Oct 2011) | 19 lines
Merged revisions 339297 via svnmerge from
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r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines
Reverting revision 333265 due to component connection problems it introduces.
I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this
problem, but first it seems prudent to remove this rather broad attempt to fix it and
instead approach this problem either from the same angle but looking only at canceling
(or possibly rescheduling) the send when we absolutely know it will cause a segfault
or, if that can't be easily accomplished, strictly from the devstate side of things.
Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe.
(issue ASTERISK-18626)
(issue ASTERISK-18078)
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r339245 | may | 2011-10-04 15:49:49 +0400 (Tue, 04 Oct 2011) | 9 lines
Merged revisions 339244 via svnmerge from
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r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines
fix forget declaration in previous change
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Review: https://reviewboard.asterisk.org/r/1445/
Is this a bug or a new feature? No responses on Asterisk-dev so I'm
committing to trunk only.
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r339148 | lmadsen | 2011-10-03 15:13:16 -0500 (Mon, 03 Oct 2011) | 14 lines
Merged revisions 339147 via svnmerge from
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r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines
Remove duplicated Maxforwards line in AMI output.
(Closes issue ASTERISK-18637)
Reported by: Jacek Konieczny
Patches:
asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny
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r339145 | lmadsen | 2011-10-03 14:55:15 -0500 (Mon, 03 Oct 2011) | 13 lines
Merged revisions 339144 via svnmerge from
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r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines
Make documentation for Dial() options 'F' and 'F()' more clear.
(Closes issue ASTERISK-18646)
Reported by: Physis Heckman
Tested by: Richard Mudgett
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r339089 | may | 2011-10-03 22:52:55 +0400 (Mon, 03 Oct 2011) | 10 lines
Merged revisions 339087 via svnmerge from
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r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines
destroy memheap mutex properly before memheap deleted
(fix memory leak occured after r304950 changes with DEBUG_THREAD compile option)
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r339088 | twilson | 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) | 17 lines
Merged revisions 339086 via svnmerge from
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r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines
Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places
After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame
is sent when a re-invite happens. If we receive a re-invite from a device
the waitstream_core was not aware of the new control frame and would drop
the call.
(closes issue ASTERISK-18610)
Reported by: Kristijan_Vrban
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r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct 2011) | 4 lines
Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here.
This function prints a list of caps instead of a hex bitfield.
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r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct 2011) | 2 lines
Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it.
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r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct 2011) | 2 lines
properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags)
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r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) | 1 line
Documentation noting the extension of CHANNEL() for chan_ooh323
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r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines
Remove the channel function OOH323() and place its options into
CHANNEL()
channel drivers should not have there own dialplan functions.
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r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) | 14 lines
Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will
turn off the gateway but the framehook is not destroyed.
this problem happens when a gateway is attempted in the dialplan and
the device is not available i may want to do fax to mail in the server
it will not be allowed.
instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id
Reverts 338904
Fix some white space.
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r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) | 8 lines
Remove T38 Gateway capability when detaching framehook.
SET(FAXOPT(gateway)=no) does not remove the capability when
detaching the framehook.
small patch to fix this problem.
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r338801 | rmudgett | 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) | 19 lines
Merged revisions 338800 via svnmerge from
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r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines
Fix segfault in analog_ss_thread() not checking ast_read() for NULL.
NOTE: The problem was reported against v1.6.2. It is unlikely to ever
happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely
to be used. The version in sig_analog.c has largely replaced it.
(closes issue ASTERISK-18648)
Reported by: Stephan Bosch
Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Stephan Bosch
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--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M channels/chan_sip.c
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r338719 | jrose | 2011-09-30 13:55:27 -0500 (Fri, 30 Sep 2011) | 9 lines
Merged revisions 338718 via svnmerge from
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r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line
Adds documentation for QueueMemberStatus event generation
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ASTERISK-17486 exposed the problem for AMI parsers.
(closes issue ASTERISK-18649)
Reported by: Jacek Konieczny
Patches:
asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny
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Review: https://reviewboard.asterisk.org/r/1463/
A small part of much larger work with DTMF duration in Asterisk,
funded by IPvision AS in Denmark.
Thanks to irroot for the review!
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r338556 | pabelanger | 2011-09-29 17:14:34 -0400 (Thu, 29 Sep 2011) | 9 lines
Merged revisions 338555 via svnmerge from
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r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines
Test modules should depend on the TEST_FRAMEWORK flag
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r338552 | qwell | 2011-09-29 15:54:55 -0500 (Thu, 29 Sep 2011) | 9 lines
Merged revisions 338551 via svnmerge from
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r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line
Test modules have a support level of core.
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r338493 | lmadsen | 2011-09-29 13:32:28 -0500 (Thu, 29 Sep 2011) | 14 lines
Merged revisions 338492 via svnmerge from
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r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines
Update documentation for SIP_HEADER.
The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated
in trunk, but not in 1.8 or 10, so I'm making them match.
(Closes issue ASTERISK-18640)
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