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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line
Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
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possible. Previously some failure cases did not result in proper FAXOPT values.
FAX-203
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274540 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/629
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines
Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(closes issue #16102)
Reported by: Delvar
Patches:
say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
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(no option for trunk, just changing the behavior)
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bitfield.
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dialplan function. Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.
FAX-202
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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r274093 | mnicholson | 2010-07-06 08:52:28 -0500 (Tue, 06 Jul 2010) | 2 lines
Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace.
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r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines
Command 'stop gracefully' doesn't.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines
If all members are paused, the wrong status is indicated.
(closes issue #17576)
Reported by: ramonpeek
Patches:
diff.txt uploaded by ramonpeek (license 266)
Tested by: ramonpeek
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force directly to the default case.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273714 f38db490-d61c-443f-a65b-d21fe96a405b
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(Also fix the typos in the comments)
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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FAX-177
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A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273427 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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(closes issue #16430)
Reported by: azbest
Tested by: azbest
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packets. This regression was introduced in r48338.
AST-359
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Also clean up some coding errors.
(closes issue #17469)
Reported by: wdoekes
Patches:
astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger
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(closes issue #17550)
Reported by: kenner
Patches:
manager.c.diff uploaded by kenner (license 1040)
Tested by: kenner
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(closes issue #17560)
Reported by: Nick_Lewis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all.
(closes issue #16029)
Reported by: Guggemand
Patches:
realtime-useragent.patch uploaded by Guggemand (license 897)
Tested by: Guggemand
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines
_Really_ skip the channel... don't just retry for another 200 cycles.
(Closes issue SWP-1652, ABE-2240)
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Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
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RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines
Don't change ownership/group/permissions on run directory, if it already exists.
(closes issue #17076)
Reported by: stuarth
Patches:
20100324__issue17076.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
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r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines
Change the way that we read include files, to accommodate for changes in GCC 4.4.
(closes issue #17472)
Reported by: seandarcy
Patches:
config2.patch uploaded by nivan (license 1066)
Tested by: nivan
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r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines
Also trim trailing blanks on #includes
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