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r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines
Remove tests for parsing address with invalid port
getaddrinfo on OS X returns with EAI_NONAME error when passed a port
greater than 65535. Linux throws no error, so remove the tests for now.
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
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r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines
The test_event unit test is occasionally failing.
Wait for the special posted event to process before adding a new
subscription.
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r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines
Remove now-useless cast of ARRAY_LEN
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r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines
Fix more ARRAY_LEN format string issues
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r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines
Cast ARRAY_LEN to size_t for ast_logging
32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast
it before using in a format string.
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r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines
[regression] Voicemail MWI is no longer sent.
When leaving a voicemail, the MWI message is never sent. The same thing
happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state correctly, but
changes to the messages in voicemail causes the light to not be set
appropriately. Very easy to reproduce.
* Made ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie values
passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002)
Reported by: lmadsen
Tested by: lmadsen, irroot
Patches:
jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621)
(closes issue ASTERISK-18019)
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r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines
Add a test to the event unit tests to catch ASTERISK-18002.
The new tests check to see if there are ANY subscribers to the event type
when ast_event_check_subscriber() is not passed any specific ie values.
(issue ASTERISK-18002)
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r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 Jun 2011) | 2 lines
Add some astdb unit tests
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r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines
Event subscription fixes.
Must commit the subscription fixes together with the integration
subscription tests. The subscription fixes cause an erroneously passing
test to fail. The new subscription tests detect errors without the
subscription fixes.
* Added missing event_names[] table entry.
* Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to
correctly detect if a subscriber exists for the proposed event.
* Made match_ie_val() and match_sub_ie_val_to_event() check the buffer
length for RAW payload types.
* Fixed error handling memory leak in ast_event_sub_activate(),
ast_event_unsubscribe(), and ast_event_queue().
* Made ast_event_new() and ast_event_check_subscriber() better protect
themselves from an invalid payload type.
* Added container lock protection between removing old cache events and
adding the new cached event in
ast_event_queue_and_cache()/event_update_cache().
* Added new event subscription tests.
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Review: https://reviewboard.asterisk.org/r/1254/
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
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Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
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r295078 | tilghman | 2010-11-15 12:30:13 -0600 (Mon, 15 Nov 2010) | 16 lines
Merged revisions 295062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r295062 | tilghman | 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) | 9 lines
Merged revisions 295026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines
Create test verifying results of expression parser
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r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct 2010) | 12 lines
Prevent multiple runs of event_sub_test from producing false failure results.
The array of test subscriptions was declared "static," meaning that the
data.count field would retain its value between runs of the test. After the
first test run, this would result in false reports of test failures.
I chose to just remove the "static" keyword from the structure since it's not
a huge deal to construct this structure during each run of the test. Another
alternative would have been to zero out the data.count fields of each test
subscription instead.
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r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) | 2 lines
More Solaris compatibility fixes
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r289581 | tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines
Solaris fixes.
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r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) | 4 lines
Solaris compatibility fixes
Review: https://reviewboard.asterisk.org/r/942/
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r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285930 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
Merged revisions 284593,284595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul 2010) | 2 lines
Remove res_crypto dependency.
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r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul 2010) | 2 lines
crypto_loaded_test depends on res_crypto, else test will fail.
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r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun, 25 Jul 2010) | 2 lines
Add trailing backslash to silence warning message.
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The ACL test was failing on Mac OS X because it would
convert the above invalid link-local address into
fe80::1234 while reporting no error from getaddrinfo().
Linux does not do this.
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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
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(closes issue #17475)
Reported by: tilghman
Review: https://reviewboard.asterisk.org/r/695/
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16461)
Reported by: skyman
Patches:
20100622__issue16461.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/737/
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tracking down the source.
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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
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pager messages).
(closes issue #14333)
Reported by: klaus3000
Patches:
20090515__issue14333.diff.txt uploaded by tilghman (license 14)
app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
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This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop. Additionally, this adds a res_timing interface, using kqueue timers.
Review: https://reviewboard.asterisk.org/r/543/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262852 f38db490-d61c-443f-a65b-d21fe96a405b
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I modified the original patch for trunk to use the unit test API.
(issue #17277)
Reported by: cappucinoking
Patches:
test_heap.diff uploaded by cappucinoking (license 1036)
Tested by: cappucinoking, russell
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This patch introduces another test in test_event.c that exercises most of the
subscription related ast_event API calls. I made some minor additions to the
existing event allocation test to increase API coverage by the test code.
Finally, I made a list in a comment of API calls not yet touched by the test
module as a to-do list for future test development.
During the development of this test code, I discovered a number of bugs in
the event API.
1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple
of different places. The API allows a subscription to all event types,
but with IE parameters, just as if it was a subscription to a specific
event type. However, the parameters were being ignored. This affected
ast_event_check_subscriber() and event distribution to subscribers.
2) Some of the logic in ast_event_check_subscriber() for checking subscriptions
against query parameters was wrong.
Review: https://reviewboard.asterisk.org/r/617/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258632 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines
Allow application options with arguments to contain parentheses, through a variety of escaping techniques.
Fixes SWP-1194 (ABE-2143).
Review: https://reviewboard.asterisk.org/r/604/
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There are two unit tests contained here.
1. "Invalid ACL" This attempts to read a bunch of badly formatted ACL entries
and add them to a host access rule. The goal of this test is to be sure that
all invalid entries are rejected as they should be.
2. "ACL" This sets up four ACLs. One is a permit all, one is a deny all, and
the other two have specific rules about which subnets are allowed and which
are not. Then a set of test addresses is used to determine whether we would
allow those addresses to access us when each ACL is applied. This test, by the
way, was what resulted in AST-2010-003's creation.
Review: https://reviewboard.asterisk.org/r/532
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pointer.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252980 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16758)
Reported by: wdoekes
Patches:
20100316__issue16758.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/561/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252849 f38db490-d61c-443f-a65b-d21fe96a405b
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Reviewboard: https://reviewboard.asterisk.org/r/554/
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