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counts so that they only get destroyed when all threads are done using
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
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r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines
Make chan_sip build under dev mode with compilers >= GCC 4.2
Thanks to jpeeler for alerting me of this
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in 12494. Due to changes in trunk to use the astobj2 i/f in the sip channel driver, the order of the entries in the config file was lost, thus the output was in a random order, but no longer.
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r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines
Fix 3 more places where failure to lock the structure could cause the wrong lock to be
unlocked. (Closes issue #12795)
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r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines
Cisco BTS sends SIP responses with a tab between the Cseq number and
SIP request method in the Cseq: header. Asterisk did not handle this
properly, but with this patch, all is well.
(closes issue #12834)
Reported by: tobias_e
Patches:
12834.patch uploaded by putnopvut (license 60)
Tested by: tobias_e
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do_monitor and creating its own thread.
Added RP-AS mwi message generation using patches from meneault as a basis.
(closes issue #8587)
Reported by: meneault
Tested by: meneault
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r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines
Port "hasvoicemail" change from SIP to other channel drivers
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r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines
People expect that if "hasvoicemail" is set in users.conf, even if "mailbox"
isn't set, that SIP will detect a mailbox.
(closes issue #12855)
Reported by: PLL
Patches:
20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: PLL
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r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines
Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP.
(closes issue #12803)
Reported by: lanzaandrea
Patches:
chan_sip.c.diff uploaded by lanzaandrea (license 496)
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r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines
Don't send a BYE on a dialog that is already gone during a REFER.
(closes issue #12865)
Reported by: flefoll
Patches:
chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244)
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function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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(closes issue #11342)
Reported by: DEA
Patches:
chan_skinny-park.txt uploaded by DEA (license 3)
chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7)
Tested by: DEA, mvanbaak
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r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines
Fix some race conditions that cause ast_assert() to report that chan_iax2 tried
to remove an entry that wasn't in the scheduler
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r121914, moving from chan_zap to chan_dahdi)
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get a reset circuit message. Fixes crash bug. (from mattf r121857, moving from chan_zap to chan_dahdi)
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines
Make calls to ast_assert() actually test something, so that the error message
printed is not nonsensical (reported by mvanbaak via #asterisk-bugs).
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get a reset circuit message. Fixes crash bug
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r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line
fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken.
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been.
(closes issue #12706)
Reported by: falves11
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
Tested by: falves11
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r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines
If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed.
(closes issue #12828)
Reported by: ramonpeek
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(Closes AST-38)
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(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
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r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
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Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
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r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines
Don't run LIST_HEAD_DESTROY on a STATIC list
(closes issue #12807)
Reported by: ys
Patches:
chan_agent_local.diff uploaded by ys (license 281)
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(closes issue #12562)
Reported by: michael-fig
Patches:
20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)
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r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line
add another LOW_MEMORY define I forgot
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r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line
only define thread storage variable if necessary for LOW_MEMORY
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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to realtime less painful in the future.
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API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
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r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) | 6 lines
If we fail to setup the PRI request channel, don't continue, exit with an error.
(closes issue #11989)
Reported by: Corydon76
Patches:
20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14)
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
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r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines
Save the callno when we're poking, because our peer structure could change
during destruction (and thus we unlock the wrong callno, causing a
cascade failure).
(closes issue #12717)
Reported by: gewfie
Patches:
20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
Tested by: gewfie
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r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines
Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox)
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines
Even of the first PING or LAGRQ doesn't get sent because it comes up too soon,
make sure to reschedule so it gets sent later.
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r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line
fixed compile issue when dev-mode is enabled
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines
Change a debug message to an actual debug message
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r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines
Merged revisions 119237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines
- Instead of only enforcing destination call number checking on an ACK, check
all full frames except for PING and LAGRQ, which may be sent by older versions
too quickly to contain the destination call number.
(As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
from being sent before the destination call number is known.
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