asterisk/asterisk.git
8 years agoAdd regression tests for issue ASTERISK-18838.
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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8 years agoThe voicemail [general] zonetag and locale variables weren't loaded
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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8 years agoDoubly linked lists unit test and update to implementation.
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed crash from orphaned MWI subscriptions in chan_sip
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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8 years agoRestore call progress code for analog ports.
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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8 years agoResolve duplicate label used in multiple priorities for the same extension.
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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8 years agoFix chan_jingle/gtalk load regression introduced in r346087
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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8 years agoFor SIP REGISTER fix domain-only URIs and domain ACL bypass.
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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8 years agoUpdate SIP MESSAGE To parsing to correctly handle URI
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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8 years agoimplement nat option for rtp channels with ooh323
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 346763 via svnmerge from
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

................
  r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines

  Merged revisions 346762 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    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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8 years agoRe-resolve the STUN address if a STUN poll fails for res_stun_monitor.
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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8 years agoChange 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.
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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8 years agoRemove the few places where we try to ast_verbose() without a newline.
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix edge case for overflow buffer.
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agor346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
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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8 years agoReverting 346525 due to accidental patch against trunk instead of 1.8
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCleaning up chan_sip/tcptls file descriptor closing.
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate queues.conf.sample documentation.
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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8 years agoFix compilation of utilities (caught by Bamboo).
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAllow each logging destination and console to have its own notion of the verbosity...
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMerged revisions 346349 via svnmerge from
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

........
  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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8 years agoFix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
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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8 years agoFix calls to ast_get_ip() not initializing the address family.
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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8 years agoMinor cleanup in chan_sip get_msg_text() function.
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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8 years agoFix ast_str_truncate signedness warning and documentation.
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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8 years agoFix res_jabber resource leaks
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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8 years agoFixed SendMessage stripping extension from To: header in SIP MESSAGE
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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8 years agoResume playing existing hold music for cached realtime MOH
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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8 years agoAdded support level for new modules
Paul Belanger [Wed, 23 Nov 2011 16:10:45 +0000 (16:10 +0000)]
Added support level for new modules
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8 years agoFix dnsmgr entries to ask for the same address family each time.
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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8 years agoClarify why the AST_LOG_* macros exist next to the LOG_* macros.
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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8 years agoAdd missing sound_only_one config variable
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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8 years agoDefault to nat=yes; warn when nat in general and peer differ
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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8 years agoAdd #tryinclude statement
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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8 years agoUpdate the documentation to better clarify how the existing commands work.
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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8 years agoFix a change in behavior in 'database show' from 1.8.
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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8 years agoAccidentally readded sipfriends.sql in r345560. This was removed
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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8 years agoAdd admin toggle mute all and participant count menu options to app_confbridge
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/

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8 years agoRemove dead code since pri_grab() can never fail.
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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8 years agoGuarantee messages go into the right folders with multiple recipients
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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8 years agoMake FastAGI HANGUP show up in AGI debug output.
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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8 years agoFix typo in sig_pri using wrong structure name.
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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8 years agoMake queue log indicate if ADDMEMBER is paused for AMI and realtime.
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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8 years agoRestore SIP DTMF overlap dialing method.
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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8 years agoFix Progress spelling error in main/pbx.c.
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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8 years agoDon't read past end of input when calling write()
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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8 years agoUpdate reqresp_parser parse_uri doxygen comments.
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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8 years agoMoves voicemail setup password entry to the end of the setup process.
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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8 years agoEnsure that a null vmexten does not cause a segfault
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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8 years agoIncreased max number of destinations.
TransNexus OSP Development [Mon, 14 Nov 2011 01:25:25 +0000 (01:25 +0000)]
Increased max number of destinations.

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8 years agomISDN Round Robin break when no channel is available
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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8 years agoDon't forget to rescan MOH files for cached realtime classes
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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8 years agoUse __alignof__ instead of sizeof for stringfield length storage.
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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8 years agoVideo format was treated as audio when removed from the file playback scheduler
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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8 years agoRemove unneeded if(params) checks in reqresp_parser.
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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8 years agoFix bad quoting of multiline mxml opaque_data that caused invalid xml.
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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8 years agoFix regression introduced by SDP fixups
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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8 years agoCheck sip.conf maxforwards parameter for range 1 <= x <= 255.
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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8 years agoMake CLI "core show channel" not hold the channel lock during console output.
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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8 years agoFix a segmentation fault when using an extension with CID matching and no CID.
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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8 years agoFix app_macro.c MODULEINFO section termination.
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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8 years agoFix potential deadlock calling ast_call() with channel locks held.
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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8 years agoMake AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.
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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8 years agoMerged revisions 344493 via svnmerge from
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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8 years agoFix another incorrect case with meetme's PIN logic and add documentation
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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8 years agoFix several bugs with SDP parsing and well-formedness of responses
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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8 years agoonly attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses
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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8 years agoFix deadlock during dialplan reload.
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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8 years agoDon't treat a host:port string as a domain
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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8 years agoAdd a unit test for ast_sockaddr_split_hostport
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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8 years agoGenerate response to Status Enquiry message with Status q.931 message.
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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8 years agoFix pin parameter behavior regression in MeetMe
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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8 years agodon't call ltohl() twice on the same value
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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8 years agoResidual changes for Asterisk v10 branch from ASTERISK-18747.
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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8 years agoFix crash when dialplan remove include is called with too few arguments.
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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8 years agoMerged revisions 343900 via svnmerge from
David Vossel [Tue, 8 Nov 2011 18:35:19 +0000 (18:35 +0000)]
Merged revisions 343900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed reference to incorrect variable if unknown host configured crash.
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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Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343852 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years agoFix boo-boo in prep_tarball script.
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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Merged revisions 343789 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake "sip show settings" CLI command get RPID flags from the right global page
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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Merged revisions 343743 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343744 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAllow built in variables to be used with dynamic weights.
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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agorespect case changes in peer names on sip reload
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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Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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8 years agoFix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid.
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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Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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8 years agoPrevent BLF subscriptions from causing deadlocks
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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Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343635 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrect the default udptl port range.
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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Merged revisions 343580 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix deadlock if peer is destroyed while sending MWI notice.
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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Merged revisions 343578 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343579 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agolist all of the codecs associated with a particular format id for CLI command "core...
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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Merged revisions 343533 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343534 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFormatting and doxygen improvements
Olle Johansson [Sun, 6 Nov 2011 09:51:09 +0000 (09:51 +0000)]
Formatting and doxygen improvements

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343492 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFinal fix memleaks in GkClient codes, same for Timer codes.
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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Merged revisions 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix sqlite config driver segfault and broken queries
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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Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343393 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343394 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove invalid flag given to iterator in func_dialgroup.c
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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Merged revisions 343336 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake room for the fax detect flags
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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Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove registertrying option in chan_sip
Terry Wilson [Wed, 2 Nov 2011 23:08:46 +0000 (23:08 +0000)]
Remove registertrying option in chan_sip

This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.

Review: https://reviewboard.asterisk.org/r/1562/
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Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix improper warning introduced by r342927 and more tweaks
Walter Doekes [Wed, 2 Nov 2011 22:46:27 +0000 (22:46 +0000)]
Fix improper warning introduced by r342927 and more tweaks

Changeset r342927 introduced a warning which was only supposed to be
emitted when a found realtime peer had an empty (or no) name. It turned
out that there were some inconsistencies left. Now found peers with an
empty name are explicitly ignored like before r342927 but better.

Reviewed by: Stefan Schmidts, Terry Wilson

Review: https://reviewboard.asterisk.org/r/1560
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Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343192 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure that string field lengths are properly aligned
Walter Doekes [Wed, 2 Nov 2011 22:02:07 +0000 (22:02 +0000)]
Ensure that string field lengths are properly aligned

Integers should always be aligned. For some platforms (ARM, SPARC) this
is more important than for others. This changeset ensures that the
string field string lengths are aligned on *all* platforms, not just on
the SPARC for which there was a workaround. It also fixes that the
length integer can be resized to 32 bits without problems if needed.

(closes issue ASTERISK-17310)
Reported by: radael, S Adrian
Reviewed by: Tzafrir Cohen, Terry Wilson
Tested by: S Adrian

Review: https://reviewboard.asterisk.org/r/1549
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Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343158 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343163 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd note about how Authenticate() application with option 'd' works.
Leif Madsen [Wed, 2 Nov 2011 19:33:49 +0000 (19:33 +0000)]
Add note about how Authenticate() application with option 'd' works.

(closes issue ASTERISK-17422)
Reported by: Leif Madsen
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Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343103 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343104 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate documentation for leastrecent strategy.
Leif Madsen [Wed, 2 Nov 2011 18:11:12 +0000 (18:11 +0000)]
Update documentation for leastrecent strategy.

In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated
to reflect how the strategy actually checks peers.

(closes issue ASTERISK-17854)
Reported by: Sebastian Denz
Patches:
     queues.conf-doc_issue.patch (License #6139)
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Merged revisions 343047 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoModify comments in MeetMe application documentation about DAHDI.
Kevin P. Fleming [Wed, 2 Nov 2011 13:46:15 +0000 (13:46 +0000)]
Modify comments in MeetMe application documentation about DAHDI.

The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
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Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSeveral fixes to the chan_sip dynamic realtime peer/user lookup
Walter Doekes [Tue, 1 Nov 2011 21:02:56 +0000 (21:02 +0000)]
Several fixes to the chan_sip dynamic realtime peer/user lookup

There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.

Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.

This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!

(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson

Review: https://reviewboard.asterisk.org/r/1395
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Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342930 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCleanup references to sipusers and sipfriends dynamic realtime families
Walter Doekes [Tue, 1 Nov 2011 19:53:26 +0000 (19:53 +0000)]
Cleanup references to sipusers and sipfriends dynamic realtime families

Somewhere between 1.4 and 1.8 the sipusers family has become completely
unused. Before that, the sipfriends family had been obsoleted in favor
of separate sipusers and sippeers families. Apparently, they have been
merged back again into a single family which is now called "sippeers".

Reviewed by: irroot, oej, pabelanger

Review: https://reviewboard.asterisk.org/r/1523
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Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 342870 from http://svn.asterisk.org/svn/asterisk/branches/10

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