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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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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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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
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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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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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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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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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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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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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verbosity level.
Review: https://reviewboard.asterisk.org/r/1599
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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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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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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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Review: https://reviewboard.asterisk.org/r/1594
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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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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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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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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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(issue ASTERISK-17973)
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(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
conf_config_parser.diff (license #5755) patch uploaded by zvision
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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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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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Review: https://reviewboard.asterisk.org/r/1593/
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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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in r342871
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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/
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Dead code makes programmers sick. I am sick of looking at it.
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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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* 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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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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* 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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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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(closes issue ASTERISK-18857)
Reported by: David M
Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
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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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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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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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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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Prevent channels been parsed repetitively.
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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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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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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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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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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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