Damien Wedhorn [Thu, 8 Dec 2011 06:59:01 +0000 (06:59 +0000)]
Fix segfault on answer.
Fix a segfault if an attempt to answer a call is made between when
the inbound call gives up (and the channel is removed) and when the
device is notified and removes the call from the device.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490
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Richard Mudgett [Wed, 7 Dec 2011 21:42:29 +0000 (21:42 +0000)]
Update AMI Getvar and Setvar documentation about supplying a channel name.
(closes issue ASTERISK-18958)
Reported by: Red
Patches:
jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
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Jonathan Rose [Wed, 7 Dec 2011 20:34:23 +0000 (20:34 +0000)]
Fix: Meetme recording variables from realtime DB use null entries over channel variables
Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.
(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
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Terry Wilson [Wed, 7 Dec 2011 20:15:29 +0000 (20:15 +0000)]
Add ASTSBINDIR to the list of configurable paths
This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.
(closes issue ASTERISK-18959)
Review: https://reviewboard.asterisk.org/r/1613/
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Richard Mudgett [Tue, 6 Dec 2011 23:58:44 +0000 (23:58 +0000)]
Make SIP INFO messages for dtmf-relay signals case insensitive.
(closes issue ASTERISK-18924)
Reported by: Kevin Taylor
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Jonathan Rose [Tue, 6 Dec 2011 22:01:00 +0000 (22:01 +0000)]
Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten
If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.
(closes issue ASTERISK-18804)
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Walter Doekes [Tue, 6 Dec 2011 20:23:13 +0000 (20:23 +0000)]
Add VM_INFO() dialplan function to gather information about a mailbox.
Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname,
language, locale, pager, password, tz.
(closes issue ASTERISK-18634)
Patch by: Kris Shaw
Review: https://reviewboard.asterisk.org/r/1568
Reviewed by: Walter Doekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192
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Walter Doekes [Tue, 6 Dec 2011 19:44:27 +0000 (19:44 +0000)]
Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still allowed.
Resolving the source address wasn't possible and yielded the string
"(null)" in SIP messages. Fixed that and a couple of not-so-correct
log messages.
(closes issue ASTERISK-18837)
Reported by: Andreas Topp
Review: https://reviewboard.asterisk.org/r/1585
Reviewed by: Matt Jordan
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Walter Doekes [Tue, 6 Dec 2011 19:30:14 +0000 (19:30 +0000)]
Add regression tests for issue ASTERISK-18838.
Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan
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Walter Doekes [Tue, 6 Dec 2011 19:28:18 +0000 (19:28 +0000)]
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.
(closes issue ASTERISK-18838)
Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
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Richard Mudgett [Tue, 6 Dec 2011 19:09:56 +0000 (19:09 +0000)]
Doubly linked lists unit test and update to implementation.
Update the doubly linked list implementation. Now safe traversing can
insert before and after the current node when traversing in either
direction.
Updated the linked lists unit test test_linkedlist to also test doubly
linked lists. The old test_dlinkedlist requires a manual check of results
and probably should be removed.
Review: https://reviewboard.asterisk.org/r/1569/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347110
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Matthew Jordan [Tue, 6 Dec 2011 17:34:35 +0000 (17:34 +0000)]
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1610/
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Richard Mudgett [Mon, 5 Dec 2011 17:44:15 +0000 (17:44 +0000)]
Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
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Jonathan Rose [Mon, 5 Dec 2011 15:04:12 +0000 (15:04 +0000)]
Resolve duplicate label used in multiple priorities for the same extension.
Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.
(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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Kinsey Moore [Mon, 5 Dec 2011 14:47:11 +0000 (14:47 +0000)]
Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy
for usage outside res_jabber. Testing of these changes focused on res_jabber
itself, so this problem was missed.
Reported-by: Michael Spiceland
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Walter Doekes [Sun, 4 Dec 2011 10:08:19 +0000 (10:08 +0000)]
For SIP REGISTER fix domain-only URIs and domain ACL bypass.
The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.
Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.
Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson
(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
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Matthew Jordan [Fri, 2 Dec 2011 23:30:21 +0000 (23:30 +0000)]
Update SIP MESSAGE To parsing to correctly handle URI
The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message. This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.
(issue ASTERISK-18903)
Review: https://reviewboard.asterisk.org/r/1597/
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Alexandr Anikin [Fri, 2 Dec 2011 19:40:21 +0000 (19:40 +0000)]
implement nat option for rtp channels with ooh323
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346816
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Alexandr Anikin [Fri, 2 Dec 2011 18:03:31 +0000 (18:03 +0000)]
Merged revisions 346763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines
Merged revisions 346762 via svnmerge from
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r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines
process null frame pointer returned by ast_rtp_instance_read correctly
(closes issue ASTERISK-16697)
Reported by: under
Patches:
segfault.diff (License #5871) patch uploaded by under
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Richard Mudgett [Thu, 1 Dec 2011 21:19:41 +0000 (21:19 +0000)]
Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.
The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change. However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.
* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.
* Fix ast_stun_request() return value consistency.
* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.
* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found. The stun_purge_socket() hack is no longer
required.
* Reduce ast_stun_request() error messages to debug output.
* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.
(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1595/
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Jonathan Rose [Thu, 1 Dec 2011 20:46:12 +0000 (20:46 +0000)]
Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.
183 is actually a session progress message.
(closes issue ASTERISK-18925)
Reported by: Sebastian Denz
Tested by: jrose
Patches:
asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139)
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Tilghman Lesher [Wed, 30 Nov 2011 23:38:34 +0000 (23:38 +0000)]
Remove the few places where we try to ast_verbose() without a newline.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655
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Tilghman Lesher [Wed, 30 Nov 2011 22:40:23 +0000 (22:40 +0000)]
Fix edge case for overflow buffer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617
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Jonathan Rose [Wed, 30 Nov 2011 22:03:02 +0000 (22:03 +0000)]
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
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Jonathan Rose [Wed, 30 Nov 2011 21:32:23 +0000 (21:32 +0000)]
Reverting 346525 due to accidental patch against trunk instead of 1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563
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Jonathan Rose [Wed, 30 Nov 2011 21:10:38 +0000 (21:10 +0000)]
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525
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Leif Madsen [Wed, 30 Nov 2011 19:37:25 +0000 (19:37 +0000)]
Update queues.conf.sample documentation.
Update the documentation surrounding the use of MONITOR_EXEC to make it more clear
that it can be used for both Monitor() and MixMonitor() usage.
(closes issue ASTERISK-17413)
Reported by: David Woolley
Patches:
issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026)
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Tilghman Lesher [Tue, 29 Nov 2011 20:32:53 +0000 (20:32 +0000)]
Fix compilation of utilities (caught by Bamboo).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346429
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Tilghman Lesher [Tue, 29 Nov 2011 18:43:16 +0000 (18:43 +0000)]
Allow each logging destination and console to have its own notion of the verbosity level.
Review: https://reviewboard.asterisk.org/r/1599
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391
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David Vossel [Tue, 29 Nov 2011 00:03:36 +0000 (00:03 +0000)]
Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines
Fixes memory leak in message API.
The ast_msg_get_var function did not properly decrement
the ref count of the var it retrieves. The way this is
implemented is a bit tricky, as we must decrement the var and then
return the var's value. As long as the documentation for the
function is followed, this will not result in a dangling pointer as
the ast_msg structure owns its own reference to the var while it
exists in the var container.
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Stefan Schmidt [Mon, 28 Nov 2011 14:34:14 +0000 (14:34 +0000)]
Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
(closes issue ASTERISK-18693)
Reported by: Davide Dal Fra
Review: https://reviewboard.asterisk.org/r/1600/
Reviewed by: Walter Doekes
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Richard Mudgett [Wed, 23 Nov 2011 23:03:32 +0000 (23:03 +0000)]
Fix calls to ast_get_ip() not initializing the address family.
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Walter Doekes [Wed, 23 Nov 2011 20:48:42 +0000 (20:48 +0000)]
Minor cleanup in chan_sip get_msg_text() function.
In r116240, get_msg_text() got an extra parameter to fix the unwanted
addition of trailing newlines to SIP MESSAGE bodies. This caused all
linefeeds to be trimmed, which isn't right either. This is a stop-gap;
the right fix is to return the original SIP request body.
Review: https://reviewboard.asterisk.org/r/1586
Reviewed by: Matt Jordan
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Walter Doekes [Wed, 23 Nov 2011 19:58:19 +0000 (19:58 +0000)]
Fix ast_str_truncate signedness warning and documentation.
Review: https://reviewboard.asterisk.org/r/1594
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Kinsey Moore [Wed, 23 Nov 2011 17:16:33 +0000 (17:16 +0000)]
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve
ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
ast_aji_get_client would sometimes bump an object's refcount and sometimes not.
Review: https://reviewboard.asterisk.org/r/1553
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Matthew Jordan [Wed, 23 Nov 2011 16:23:34 +0000 (16:23 +0000)]
Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address. In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting. This patch adds the full URI passed in
from the message core to the sip_pvt structure.
(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1597/
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Terry Wilson [Wed, 23 Nov 2011 16:12:34 +0000 (16:12 +0000)]
Resume playing existing hold music for cached realtime MOH
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer
properly resumes playing back a file between different holds in the same call.
This is because scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the filename
matched the pointer to the filename in a particular position in the array. An
easy fix is to save the filename instead of a pointer to it and then do a
strcmp instead of comparing the addresses.
(closes issue ASTERISK-18912)
Review: https://reviewboard.asterisk.org/r/1596/
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Paul Belanger [Wed, 23 Nov 2011 16:10:45 +0000 (16:10 +0000)]
Added support level for new modules
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Richard Mudgett [Tue, 22 Nov 2011 23:06:11 +0000 (23:06 +0000)]
Fix dnsmgr entries to ask for the same address family each time.
The dnsmgr refresh would always get the first address found regardless of
the original address family requested. So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.
* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
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Walter Doekes [Tue, 22 Nov 2011 20:32:51 +0000 (20:32 +0000)]
Clarify why the AST_LOG_* macros exist next to the LOG_* macros.
(issue ASTERISK-17973)
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Paul Belanger [Tue, 22 Nov 2011 16:41:58 +0000 (16:41 +0000)]
Add missing sound_only_one config variable
(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
conf_config_parser.diff (license #5755) patch uploaded by zvision
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Terry Wilson [Mon, 21 Nov 2011 21:09:59 +0000 (21:09 +0000)]
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer
nat settings differ in whether to respond to the port a request is sent
from or the port listed for responses in the Via header. In 1.4 and 1.6.2,
this would mean if one setting was nat=yes or nat=route and the other was
either nat=no or nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no.
In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all possible.
For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
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Paul Belanger [Mon, 21 Nov 2011 16:40:17 +0000 (16:40 +0000)]
Add #tryinclude statement
This provides the same functionality as #include however an asterisk module will
still load if the filename does not exist.
Review: https://reviewboard.asterisk.org/r/1476/
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Tilghman Lesher [Sat, 19 Nov 2011 15:11:45 +0000 (15:11 +0000)]
Update the documentation to better clarify how the existing commands work.
Review: https://reviewboard.asterisk.org/r/1593/
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Tilghman Lesher [Fri, 18 Nov 2011 22:20:47 +0000 (22:20 +0000)]
Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record. Until this patch, this
did not work for the full key.
Closes issue ASTERISK-18886
Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
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Matthew Jordan [Thu, 17 Nov 2011 19:47:29 +0000 (19:47 +0000)]
Accidentally readded sipfriends.sql in r345560. This was removed
in r342871
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Matthew Jordan [Thu, 17 Nov 2011 18:09:13 +0000 (18:09 +0000)]
Add admin toggle mute all and participant count menu options to app_confbridge
This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560
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Richard Mudgett [Thu, 17 Nov 2011 17:31:16 +0000 (17:31 +0000)]
Remove dead code since pri_grab() can never fail.
Dead code makes programmers sick. I am sick of looking at it.
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Jonathan Rose [Wed, 16 Nov 2011 14:56:03 +0000 (14:56 +0000)]
Guarantee messages go into the right folders with multiple recipients
Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.
(closes issue ASTERISK-18245)
Reported by: Matt Jordan
(closes issue ASTERISK-18246)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1589/
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Richard Mudgett [Tue, 15 Nov 2011 20:11:06 +0000 (20:11 +0000)]
Make FastAGI HANGUP show up in AGI debug output.
* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.
(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
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Richard Mudgett [Tue, 15 Nov 2011 18:18:11 +0000 (18:18 +0000)]
Fix typo in sig_pri using wrong structure name.
It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.
(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
sig_pri.c.diff (License #5755) patch uploaded by zvision
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Richard Mudgett [Mon, 14 Nov 2011 22:27:42 +0000 (22:27 +0000)]
Make queue log indicate if ADDMEMBER is paused for AMI and realtime.
* Add parameter to queue log ADDMEMBER to indicate if the member is
paused.
(closes issue ASTERISK-18645)
Reported by: garlew
Patches:
paused.diff (License #5337) patch uploaded by garlew
Tested by: rmudgett, garlew
Review: https://reviewboard.asterisk.org/r/1469/
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Richard Mudgett [Mon, 14 Nov 2011 22:05:39 +0000 (22:05 +0000)]
Restore SIP DTMF overlap dialing method.
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.
See ASTERISK-18702 it has a very good description of the issue.
I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.
* Added 'dtmf' enum value to sip.conf allowoverlap config option. The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.
* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.
* Fixed get_destination() inconsistency with the pickup extension
matching.
* Fixed initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702)
Reported by: Pavel Troller
Review: https://reviewboard.asterisk.org/r/1517/
Review: https://reviewboard.asterisk.org/r/1582/
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Richard Mudgett [Mon, 14 Nov 2011 20:48:19 +0000 (20:48 +0000)]
Fix Progress spelling error in main/pbx.c.
(closes issue ASTERISK-18857)
Reported by: David M
Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
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Terry Wilson [Mon, 14 Nov 2011 19:12:49 +0000 (19:12 +0000)]
Don't read past end of input when calling write()
int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/
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Walter Doekes [Mon, 14 Nov 2011 19:03:29 +0000 (19:03 +0000)]
Update reqresp_parser parse_uri doxygen comments.
The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.
(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
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Jonathan Rose [Mon, 14 Nov 2011 16:21:06 +0000 (16:21 +0000)]
Moves voicemail setup password entry to the end of the setup process.
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.
(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
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Kinsey Moore [Mon, 14 Nov 2011 15:11:09 +0000 (15:11 +0000)]
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null. This change handles that situation to avoid
a segfault.
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TransNexus OSP Development [Mon, 14 Nov 2011 01:25:25 +0000 (01:25 +0000)]
Increased max number of destinations.
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Gregory Nietsky [Sat, 12 Nov 2011 16:32:45 +0000 (16:32 +0000)]
mISDN Round Robin break when no channel is available
Prevent channels been parsed repetitively.
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Terry Wilson [Sat, 12 Nov 2011 00:36:37 +0000 (00:36 +0000)]
Don't forget to rescan MOH files for cached realtime classes
Realtime MOH class caching was implemented because without it, you would build
a completely new MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this broke re-scanning
for file changes for realtime MOH classes. This patch corrects that issue.
(closes issue ASTERISK-18039)
Review: https://reviewboard.asterisk.org/r/1579/
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Walter Doekes [Fri, 11 Nov 2011 22:00:14 +0000 (22:00 +0000)]
Use __alignof__ instead of sizeof for stringfield length storage.
Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.
Review: https://reviewboard.asterisk.org/r/1573
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Matthew Jordan [Fri, 11 Nov 2011 21:57:46 +0000 (21:57 +0000)]
Video format was treated as audio when removed from the file playback scheduler
This patch fixes the format type check in ast_closestream and
filestream_destructor. Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to filestream_close.
(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1580/
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Walter Doekes [Fri, 11 Nov 2011 21:37:53 +0000 (21:37 +0000)]
Remove unneeded if(params) checks in reqresp_parser.
Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/
for no apparent reason. There is no way that params could become NULL in
that piece of code, so I removed these excess checks again.
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Walter Doekes [Fri, 11 Nov 2011 21:33:54 +0000 (21:33 +0000)]
Fix bad quoting of multiline mxml opaque_data that caused invalid xml.
The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.
(closes issue ASTERISK-18852)
Reported by: peep_ on IRC
Review: https://reviewboard.asterisk.org/r/1577
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Kinsey Moore [Fri, 11 Nov 2011 20:15:16 +0000 (20:15 +0000)]
Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails. Make sure capability is only touched if RTP is active. This
regression was introduced in R344385.
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Richard Mudgett [Fri, 11 Nov 2011 18:37:32 +0000 (18:37 +0000)]
Check sip.conf maxforwards parameter for range 1 <= x <= 255.
JIRA AST-710
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Richard Mudgett [Fri, 11 Nov 2011 18:02:52 +0000 (18:02 +0000)]
Make CLI "core show channel" not hold the channel lock during console output.
Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system. It could block the system if the
console output is halted or paused.
* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.
* Removed use of C++ keyword as a variable name. out renamed to obuf.
* Checked allocation of obuf for failure so will not crash.
(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
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Jonathan Rose [Fri, 11 Nov 2011 15:47:39 +0000 (15:47 +0000)]
Fix a segmentation fault when using an extension with CID matching and no CID.
Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.
(closes issue ASTERISK-18392
Reported By: Ales Zelenik
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Richard Mudgett [Thu, 10 Nov 2011 23:21:30 +0000 (23:21 +0000)]
Fix app_macro.c MODULEINFO section termination.
(closes issue ASTERISK-18848)
Reported by: Tony Mountifield
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Richard Mudgett [Thu, 10 Nov 2011 23:02:46 +0000 (23:02 +0000)]
Fix potential deadlock calling ast_call() with channel locks held.
Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks
held. Chan_local attempts to do deadlock avoidance in its ast_call()
callback and could deadlock if a channel lock is already held.
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Richard Mudgett [Thu, 10 Nov 2011 22:38:29 +0000 (22:38 +0000)]
Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.
It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line support was
added, this information was always the same at this point.
(closes issue ASTERISK-18152)
Reported by: Thomas Farnham
Tested by: rmudgett
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David Vossel [Thu, 10 Nov 2011 21:56:16 +0000 (21:56 +0000)]
Merged revisions 344493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines
Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
When a conference user enters the DTMF menu they are suspended from the
bridge while the channel is handed off to the DTMF feature code. If a
user entered this state and hungup, there existed a race condition where
the channel could not exit the conference because it was waiting on a
signal that would never arrive. This patch fixes that, because it would
stupid for me to talk about the problem and commit a patch for something else.
(closes issue ASTERISK-18829)
Reported by: zvision
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Kinsey Moore [Thu, 10 Nov 2011 21:15:39 +0000 (21:15 +0000)]
Fix another incorrect case with meetme's PIN logic and add documentation
This fixes an issue where a user of a dynamic conference was asked for a PIN
twice. This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.
(closes issue AST-670)
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Kinsey Moore [Thu, 10 Nov 2011 18:15:02 +0000 (18:15 +0000)]
Fix several bugs with SDP parsing and well-formedness of responses
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
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Matthew Nicholson [Thu, 10 Nov 2011 16:29:13 +0000 (16:29 +0000)]
only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses
Patch by: jkonieczny (modified)
ASTERISK-18490
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Richard Mudgett [Wed, 9 Nov 2011 20:55:43 +0000 (20:55 +0000)]
Fix deadlock during dialplan reload.
Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
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Terry Wilson [Wed, 9 Nov 2011 20:10:52 +0000 (20:10 +0000)]
Don't treat a host:port string as a domain
The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/
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Terry Wilson [Wed, 9 Nov 2011 19:31:27 +0000 (19:31 +0000)]
Add a unit test for ast_sockaddr_split_hostport
Review: https://reviewboard.asterisk.org/r/1575/
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Alexandr Anikin [Wed, 9 Nov 2011 19:08:44 +0000 (19:08 +0000)]
Generate response to Status Enquiry message with Status q.931 message.
Some PBXes require this for call status checking
(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
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Kinsey Moore [Wed, 9 Nov 2011 17:15:44 +0000 (17:15 +0000)]
Fix pin parameter behavior regression in MeetMe
The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.
(closes issue ASTERISK-18488)
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Matthew Nicholson [Wed, 9 Nov 2011 15:28:33 +0000 (15:28 +0000)]
don't call ltohl() twice on the same value
ASTERISK-18739
Patch by: pawel (modified)
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Richard Mudgett [Tue, 8 Nov 2011 22:14:38 +0000 (22:14 +0000)]
Residual changes for Asterisk v10 branch from ASTERISK-18747.
Residual changes for Asterisk v10 branch from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs
callid hash key change fix.
* Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from
dialogs_rtpcheck. This is an optimization to avoid an unneeded
lock/unlock and object search when using ao2_unlink.
* Prevent crash in check_rtp_timeout() if dialog->rtp is NULL.
Review: https://reviewboard.asterisk.org/r/1557/
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Walter Doekes [Tue, 8 Nov 2011 19:29:25 +0000 (19:29 +0000)]
Fix crash when dialplan remove include is called with too few arguments.
"dialplan remove include x from y" crashed when the amount of arguments
was less than 6.
(closes issue ASTERISK-18762)
Reported by: Andrey Solovyev
Tested by: Andrey Solovyev
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David Vossel [Tue, 8 Nov 2011 18:35:19 +0000 (18:35 +0000)]
Merged revisions 343900 via svnmerge from
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r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
Fixes regression caused by r343635
There was a missing unlock for a function return that is only
present in Asterisk 10 and Asterisk Trunk.
(closes issue ASTERISK-18839)
Reported by: Michael L. Young
Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
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Richard Mudgett [Tue, 8 Nov 2011 18:02:51 +0000 (18:02 +0000)]
Fixed reference to incorrect variable if unknown host configured crash.
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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Leif Madsen [Tue, 8 Nov 2011 13:23:27 +0000 (13:23 +0000)]
Fix boo-boo in prep_tarball script.
A hardcoded a branch number was in the prep_tarball which could not work. Changed
it to the variable.
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Kinsey Moore [Mon, 7 Nov 2011 22:37:51 +0000 (22:37 +0000)]
Make "sip show settings" CLI command get RPID flags from the right global page
The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command
pulled the flags from the incorrect global flags page. These are now read from
sip global flags page 0.
(closes issue AST-711)
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Leif Madsen [Mon, 7 Nov 2011 21:58:14 +0000 (21:58 +0000)]
Allow built in variables to be used with dynamic weights.
You can now use the built in variables , , and
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.
(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)
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Matthew Nicholson [Mon, 7 Nov 2011 21:44:05 +0000 (21:44 +0000)]
respect case changes in peer names on sip reload
ASTERISK-18669
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Richard Mudgett [Mon, 7 Nov 2011 21:29:01 +0000 (21:29 +0000)]
Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid.
Changing an object value used as a container key requires removing the
object from the container and reinserting it.
* Created change_callid_pvt() to call instead of build_callid_pvt(). The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
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Kinsey Moore [Mon, 7 Nov 2011 20:35:58 +0000 (20:35 +0000)]
Prevent BLF subscriptions from causing deadlocks
Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks.
This function now requires that both the peer and associated pvt be unlocked
before it is called for cases where peer and peer->mwipvt form a circular
reference.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
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Walter Doekes [Mon, 7 Nov 2011 19:58:44 +0000 (19:58 +0000)]
Correct the default udptl port range.
The udptl port range was defined as 4000-4999 in the udptl.conf.sample,
as 4500-4599 if you didn't have a config and 4500-4999 if your config
was broken. Default is now 4000-4999.
(closes issue ASTERISK-16250)
Reviewed by: Tilghman Lesher
Review: https://reviewboard.asterisk.org/r/1565
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Richard Mudgett [Mon, 7 Nov 2011 19:54:09 +0000 (19:54 +0000)]
Fix deadlock if peer is destroyed while sending MWI notice.
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
NOTE: This particular deadlock will not happen with Asterisk 10, but some
of the changes still apply.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
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Matthew Nicholson [Mon, 7 Nov 2011 18:42:04 +0000 (18:42 +0000)]
list all of the codecs associated with a particular format id for CLI command "core show codec"
AST-699
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Olle Johansson [Sun, 6 Nov 2011 09:51:09 +0000 (09:51 +0000)]
Formatting and doxygen improvements
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Alexandr Anikin [Fri, 4 Nov 2011 19:50:10 +0000 (19:50 +0000)]
Final fix memleaks in GkClient codes, same for Timer codes.
(these memleaks stop development of gk codes, now i can continue)
Fix printHandler 'Unbalanced Structure' issues with locking printHandler
data for single thread.
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Walter Doekes [Thu, 3 Nov 2011 20:37:50 +0000 (20:37 +0000)]
Fix sqlite config driver segfault and broken queries
The sqlite realtime handler assumed you had a static config configured
as well. The realtime multientry handler assumed that you weren't using
dynamic realtime.
(closes issue ASTERISK-18354)
(closes issue ASTERISK-18355)
Review: https://reviewboard.asterisk.org/r/1561
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Richard Mudgett [Thu, 3 Nov 2011 19:57:49 +0000 (19:57 +0000)]
Remove invalid flag given to iterator in func_dialgroup.c
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Terry Wilson [Thu, 3 Nov 2011 15:40:49 +0000 (15:40 +0000)]
Make room for the fax detect flags
The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
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projects / asterisk/asterisk.git / log