This should fix all the CDR tests that were not passing. When they would
originate a call, all fields in the INVITE that contained the source port would
have the port set to 0. Most troubling of these was the Contact header. Tests
are passing locally now and should also pass on the bamboo build agents.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Part of the change with the IPv6 changes is to treat a host:port as
a single 'domain' entity. This test was not updated to have the correct
expectation after calling parse_uri().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
warning (at least with gcc 4.4.4):
netsock2.c:492: warning: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
So we're back to using memcpy()...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
